[asterisk-users] Sending SIP 183 Session Progressing
i'm not sure if this is a -users or a -dev question, but am sending it here anyways. discussion could move to -dev if chan_sip.c code needs to be amended/explained. first up, all this on asterisk 1.2.10 on freebsd 6.1. here's the beef: from a particular sip softphone we're playing with, we notice that calls to another SIP phone (same LAN) result in the /lack/ of a ringing tone on the softphone. however, calls from the same softphone to a PSTN/Mobile number (through a TE405P) result in proper behaviour on the softphone with a ringing tone. an ethereal trace of both types of calls results in only one difference. for calls to the PSTN/Mobile thru libpri/chan_zap, asterisk returns a SIP 183 Session Progress[1] packet in between the 100 Trying and 180 Ringing, while for calls from the softphone to another SIP phone it's 100 Trying followed immediately by 180 Ringing. so my question is, is the softphone behaving correctly in not playing a ringing tone to the user without the 183 packet inspite of the 180 Ringing packet being received ? alternatively, since we aren't able to change the softphone, will i break anything big if i force asterisk to send the 183 packet immediately after sending the 100 Trying packet in sip_indicate() ? alternatively, in reading the RFCs, i came across RFC3398 which speficies mappings between ISDN Cause Codes and SIP responses. has this mapping been implemented in asterisk at the moment, either in 1.2 or the upcoming 1.4 ? [1] the 183 Session Progress packet is triggered by the receipt of a PRI PROGRESS indicator from libpri, which gets translated to a AST_CONTROL_PROGRESS and thence a 183 Session Progress to SIP. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.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
[asterisk-users] Re: Asterisk load testing
Hi, Am Dienstag, 15. August 2006 00:28 schrieb [EMAIL PROTECTED]: Hi, did anyone try do load-testing on asterisk, for sip channel calls? I want to have a rough estimate about - how many calls, an asterisk server, running on say dual 240 opteron with 1 GB memory, can handle? Also how much internet bandwidth does a typical call requires? I heard around 20Kbps with typical codecs, is that right? we have been responsible for an Asterisk server ( Celeron 2 GHz, 256 MB) that was treated with the ABACUS 5000. More info on the ABACUS: http://www.spirentcom.com/analysis/technology.cfm?az-c=plmedia=7ws=325ss=111 The ABACUS simulated up to 1100 SIP clients with 550 SIP calls between these clients. I'm still waiting for a more detailed report from the consultant who operated the ABACUS. Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice over IP - Lösungen ___ --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] SPA-942 TFTP Provisioning
Jeremiah Millay wrote: I'm trying to provision some spa-942 phones via TFTP. The phones get their address from a dhcp server which sends it option 66 (address of the tftp server). After spending some time with the phones and even breaking down to sniff traffic from the phones I see that they are not requesting their config from tftp. I can kind of fake the phones into grabbing their configs by doing something like: Make sure you reset to factory default those phones. Quite possible you've disabled resync on reboot or something like that. Our SPA-941 are resyncing from dhcp ok. HTH, Vahan ___ --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] Problems with incoming authentication
Hi,I am Chandra from India. We have Installed Asterisk in our organization. We want to buy a VoIP plan to make calls to US. I have some doubts. Please clarify. 1) How is the Voicepulse service?2) Is Voicepulse working fine with Asterisk?3) Can I configure Voicepulse easily with Asterisk?4) From Teliax and Voicepulse, Which is offering better service?Looking forward to your response. Thank you.Regards,Chandra.David Freeman [EMAIL PROTECTED] wrote: Sometimes I can receive a call to my DID, but sometimes it just rings and rings and I see these messages in the full log:[Aug 14 23:32:12] DEBUG[3556] res_crypto.c: Key failed verification: voicepulse20060419[Aug 14 23:32:12] NOTICE[3556] chan_iax2.c: Host 64.61.93.87 failed to authenticate as voicepulse[Aug 14 23:32:13] DEBUG[3550] res_crypto.c: Key failed verification: voicepulse20060419[Aug 14 23:32:13] NOTICE[3550] chan_iax2.c: Host 64.61.93.90 failed to authenticate as voicepulseUsually, when this happens, I can immediately re-dial the DID and * receives the call and sends it on the dialplan.I've tried configuring my IAX2 user details to no use rsa and no key, but the problem persists. I've tried to delete my IAX2 trunks to just use the SIP ones and I get the same problem...in fact, if I only have SIP trunks, I can't receive any calls to the DID.I'm using Asterisk SVN-trunk-r39753M currently, updated today. Any help would be appreciated, I'm running out of ideas. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Stay in the know. Pulse on the new Yahoo.com. Check it out. Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail Beta.___ --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] Unable to receive Incoming calls to my DID. Please tell me the solution
Hi,Thank you very much for your patience to give solutions for me. Today is holiday for us because of our Independance day. Tomorrow I will do and check as suggested by you and let you know. Once again, Thank you.Regards,Chandra.Lacy Moore - Aspendora [EMAIL PROTECTED] wrote: I struggled with one provider for a long time until finally realizing my username on their site was not my username that I was supposed to be using in sip configuration. Make sure you are using the right username and password. However, it would seem that you would not be able to make an outgoing call using the wrong username/password combination. One thing I have not seen in your posts is your firewall information. Your firewall may be setup to allow outgoing connections, but not incoming. I would not depend on info from a provider. You may very well be registering with them, but your firewall may be blocking the incoming call. If you think you have no firewall, check again. IPTABLES might have loaded itself and it may be blocking. Try: service iptables stop and then try the incoming call again. I've been burned twice due to this. Something has changed in the way I configure my linux boxes, and for some reason iptables is starting. On 8/14/06, Rich Adamson [EMAIL PROTECTED] wrote: Crazy Boy wrote: Hi, Thank you for your response. As you said, I executed the command "sip show registry". But, its not showing anything. Teliax people are also telling that my Asterisk server doesn't register with Teliax. So, the final conclusion is "My Asterisk server doesn't register with Teliax". Here I am giving my configuration files. Now, What I have to do to register my Asterisk server with Teliax? Please tell me. SIP.CONF contents: [general] register = xyz.abc:[EMAIL PROTECTED] [authentication] auth =xyz.abc:[EMAIL PROTECTED] Double check the above two statements to ensure the userid and passwordare exactly those provided to you by teliax. There is nothing else inyour config that impacts the register statement with the exception of nat'ing.It would appear from your other config statements that asterisk might belocated behind a firewall or nat box. If so, read the documentation onthat, and look at the asterisk/configs/sip.conf.sample file. Specifically the section on "NAT SUPPORT".You might also want to read more about using the diagnostic toolsavailable to you within asterisk. Setting verbose and/or debug to a highnumber and copy/paste the CLI output associated with the problem. Or, start using the CLI with something like:asterisk -rvv [teliax-incoming] exten = 3031234567, 1, Answer() exten = 3031234567, 2, Dial(SIP/105,15) The above has nothing to do with registering with teliax, but you do notwant to "answer" a call before ringing the sip phone. Take thatstatement out of there. When the sip phone answers an incoming call, asterisk will automatically send the answer to teliax.___--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 -- Lacy MooreAspendora, Inc. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Next-gen email? Have it all with the all-new Yahoo! Mail Beta.___ --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] SIP Qualify
Hello, is the same qualify behaviour with iax? i.e. iax qualify also pools peers every 60s and wait qualify=xx ms to peer respond? I have set for iax peer: Qualify: every 5000ms when OK, every 1ms when UNREACHABLE (sample smoothing On) so, I accept 5s delay in response from client, but asterisk logs still this messages (look at measured delay cca 2000ms vs. 5000ms, that I can accept) I'm confused :-\ PJ Aug 15 02:26:32 NOTICE[28564] chan_iax2.c: Peer 'wilder' is now TOO LAGGED (2023 ms)! Aug 15 02:27:04 NOTICE[28564] chan_iax2.c: Peer 'wilder' is now TOO LAGGED (2026 ms)! Aug 15 02:31:18 NOTICE[28564] chan_iax2.c: Peer 'wilder' is now TOO LAGGED (2034 ms)! Aug 15 02:31:40 NOTICE[28564] chan_iax2.c: Peer 'wilder' is now TOO LAGGED (2025 ms)! Aug 15 02:33:52 NOTICE[28564] chan_iax2.c: Peer 'wilder' is now TOO LAGGED (2027 ms)! Aug 15 07:08:53 NOTICE[28564] chan_iax2.c: Peer 'prec' is now TOO LAGGED (2051 ms)! Aug 15 08:28:45 NOTICE[28564] chan_iax2.c: Peer 'prec' is now TOO LAGGED (2017 ms)! Alexander Lopez wrote: Qualify does what the name implies qualifies the connection' It pools every 60s but it calculates he time it took for the packet to reach the end device. If the endpoint has a latentcy than the qualify parameter, * considers the endpoint unreachable. This does not however address the point you made in another post about RINGING before the INVITE. It is still possible to have a phone go dead in the 60sec between qualify re-checks. There are several post in history about qualify and it sending LARGE amounts of traffic to endpoints. I think it was John Todd that was the OP on the subject IIRC. SNIP ___ --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] More SNOM, Message Indicator/Retrieval issues
The D_key F_key options are AFAIK only available on the 6.X snom firmware.
I have Application-Version:snom360-SIP 6.2.2
the funcionality is great, even the xml minibrowser works correctly for
menus and images.
Overall sound quality seems better than the 5.x images as well
Bails
J. Oquendo wrote:
Sorry but I don't follow you. Where in the configuration is there a
dkey_retrieve option? I've never seen it and I checked preferences,
function keys, speed dials, etc. Can't find this option.
If you mean line configuration, this is how I have it set up
Configuration Line 1
Login Information:
Line active:(X) on ( ) off
Displayname:Firstname Lastname
Account:1230
Password:
Registrar:192.168.1.91
Authentication Username:1230
Mailbox:[EMAIL PROTECTED]
We have voicemail set to work on 00. So when someone calls in from the
outside it works fine:
InboundCallFromOutside -- AutoAttendant -- 00 -- Comedian Mail --
Mailbox -- Password -- You have x messages
But internally, when we hit the retrieve button we get:
Retrieve Button -- Comedian Mail -- Mailbox -- Password -- incorrect
login
If the user hangs up then presses 00 from that same phone they get:
Comedian Mail -- Mailbox -- Password -- You have x messages
This is all I have for voicemail in extensions.conf
extensions.conf:exten = 00,1,VoicemailMain([EMAIL PROTECTED])
exten = *1230,1,Goto(gateway,${EXTEN:1},2)
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[asterisk-users] PRI Clock Signal Problem
I am using a Digium TE110P to connect to my local telco's PRI network. The problem is that I do not pick up a clock signal from the telco. According to zttool and /proc/zaptel/1 the sync source is 'internally clocked'. By not using the telco's clock source I'm having problems with faxes and occasional HDLC errors and dropped calls. /etc/zaptel.conf looks like this: span=1,1,0,ccs,hdb3 dchan=16 bchan=1-15,17-31 I suspected the PRI line may be faulty so I moved the server to another location where I know I'm picking up a clock source on a 4 port Digium card. The problem persisted on the single port TE110P. I then replaced the TE110P with an identical TE110P and the new card still uses the internal clock. I even tried 3 different motherboards (2 Intel and 1 Gigabyte) but nothing changed. Did anyone have a similar problem? ___ --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] Is anybody moderating this list?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Zeeshan Zakaria wrote: Hi, It doesn't seem that anybody is moderating digium's mailing lists, thats why some uncivilized people with no manners to talk keep making this list dirty. Recently I've noticed increase in irresponsibly typed and rudly answered messages. If there are moderators here, they should stop it and kick these people out of these mailing lists. If someone has broken the lists policy post a mail to the list owner. Some people have been banned for posting commercial mails to the list or for spamming, but not being friendly is hardly something to moderate for. If you don't like someone's response, don't read it. - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE4Z4AS6d5vy0jeVcRAsynAJ9DkkDck1MdNcDqhnAFQAQ17eVL6wCfRnj/ lINZG4VpdJcJ/DDFjqIrzoM= =7iDZ -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] Run As User Asterisk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Forrest Beck wrote: Does anyone have a listing on file/directories that asterisk needs ownership of to run as a user other than root? I know about the major items --- /etc/asterisk, /var/spool/asterisk/, /var/lib/asterisk, etc... Anyone have a script to fix all the directories? vi /etc/asterisk/asterisk.conf maybe? - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE4Z4fS6d5vy0jeVcRAt8VAJ9nOO2tYL3nGavDzs8GJHuyKxIn9gCeMx3V BoTXDsieNyGL7p3nmEuoHBU= =ci5J -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] SIP Qualify
Only the Asterisk box that a phone is registered on WILL send the sip notify messages. The others will have no idea where to send them, and will not do so. - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Tuesday, August 15, 2006 12:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] SIP Qualify Yes, it might be a problem in our situation. We have three Asterisk boxes in a 'cluster'. The sip.conf is identical on all three. In that case, all three of the Asterisk boxes in our cluster are going to send sip options messages to the phones, which is silly. Only the Asterisk box that a phone is registered on needs to send the sip notify messages. The rest are a waste. I'm not sure how we'd work around this. We may just have to make do with the caller of an unavailable phone getting ringback until the dial timeout occurs. Doug. -Original Message- From: Alexander Lopez [mailto:[EMAIL PROTECTED] Sent: Mon 8/14/2006 9:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: RE: [asterisk-users] SIP Qualify Qualify does what the name implies qualifies the connection' It pools every 60s but it calculates he time it took for the packet to reach the end device. If the endpoint has a latentcy than the qualify parameter, * considers the endpoint unreachable. This does not however address the point you made in another post about RINGING before the INVITE. It is still possible to have a phone go dead in the 60sec between qualify re-checks. There are several post in history about qualify and it sending LARGE amounts of traffic to endpoints. I think it was John Todd that was the OP on the subject IIRC. SNIP ___ --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 The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy 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] Abstraction for a newbie
Dominic,I am not familiar with Trixbox, however a similar service from Sipgate in the UK is configured in Asterisk quite easily. Check out this link. http://www.sipgate.co.uk/faq/index.php?aktion=artikelrubrik=715id=540lang=dehighlight=asteriskFrom what I understand, these type of 'trunks' are really just SIP accounts. Having Asterisk or Trixbox register the account gives you control of how to use the line, as opposed to configuring a SIP account on a telephone. Hope this helps,ColinOn 8/14/06, Dominic Son [EMAIL PROTECTED] wrote: Thank you Mark. I've went from The number you are dialing is not in service, please check the number and dial again to a fast busy tone...I think I'm getting closer..-- Anything else, let me know. -Dominic Sonwww.DominicSon.com On 8/12/06, Mark Phillips [EMAIL PROTECTED] wrote:Sounds to me like you don't have a proper connection with Stanaphone. The only time you'll get these problems is when they cannot contact youto forward the call to your system.Double check you firewall settings. They need to be able to reach yoursystem on port 5060UDP (assuming SIP) as well as ports 1-2UDP (Asterisk default media ports).They'll contact yo when a call comes in. You'll accept the call and atthe same time tell them which port to send the incoming audio to.They'll also tell you where to send your outgoing audio. Hope that helps.MarkOn Fri, 2006-08-11 at 15:45 -0700, Dominic Son wrote: Hi. Can someone explain to a right brained person what is going on with In/out bound trunks, how it connects to my Trixbox.. 1. i get issued a free NY phone number from a voip service like stanaphone . 2. i then call this number, it connects to the stanaphone voicemail 3. i turn off the voicemail because i want it to connect to my Askterisk, I've set up all the trunks in the PBX setup, ( sip.stanaphone, etc) 4. now i call my NY number, and it says 'this phone is not in service, please check the number and dial again' my Q: how does this work, more specifically, if i turned off the VM, how does stanaphone then know to look for my asterisk server to use the trixbox? -- Anything else, let me know. ___ --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 listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--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
Re: [asterisk-users] Abstraction for a newbie
Dominic Son wrote: Hi. Can someone explain to a right brained person what is going on with In/out bound trunks, how it connects to my Trixbox.. 1. i get issued a free NY phone number from a voip service like stanaphone . 2. i then call this number, it connects to the stanaphone voicemail 3. i turn off the voicemail because i want it to connect to my Askterisk, I've set up all the trunks in the PBX setup, ( sip.stanaphone, etc) 4. now i call my NY number, and it says 'this phone is not in service, please check the number and dial again' my Q: how does this work, more specifically, if i turned off the VM, how does stanaphone then know to look for my asterisk server to use the trixbox? I had a similar problem. Turned out the extension was not correctly configured. If you check the console with asterisk -r and run with debug/verbosity up around 5 you will see that the call is hitting your asterisk box, but asterisk doesn't know what to do with it. Make sure the extension is correctly configured in extensions.conf, reload and try again. Hope that helps. Regards, Austin. signature.asc Description: OpenPGP digital 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] Run As User Asterisk
On Mon, Aug 14, 2006 at 09:44:39PM -0400, Forrest Beck wrote: Does anyone have a listing on file/directories that asterisk needs ownership of to run as a user other than root? I know about the major items --- /etc/asterisk, /var/spool/asterisk/, /var/lib/asterisk, etc... Anyone have a script to fix all the directories? /etc/asterisk needs to be readable to asterisk. Except voicemail.conf (or if you wish to allow asterisk to re-write extensions.conf). As it has various files with passwords, it is generally a good idea to make it non-accessible to others. Generally /var/*/asterisk (/var/log/asterisk , /var/lib/asterisk , /var/run/asterisk) should be of the asterisk user. You can use some finer tuning. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.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] Issues compiling addons on Fedora Core 3
Still having no luck on this one. Have searched ye olde Google, and found a couple of other references to the same error, but stemming back to an earlier revision which was supposedly 'fixed.' No one's seen this before and knows what causes it? Newer version of gcc perhaps? GCC compiled differently? A particular version of the MySQL libs? N. On Sat, 12 Aug 2006 13:48:40 -0400, sip wrote I have zero problem compiling the addons 1.23 on FC4 and RH4, but for some reason, when I try to compile them on FC3, I get this: cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:164:36: macro AST_LIST_REMOVE requires 4 arguments, but only 3 given app_addon_sql_mysql.c: In function `del_identifier': app_addon_sql_mysql.c:164: error: `AST_LIST_REMOVE' undeclared (first use in this function) app_addon_sql_mysql.c:164: error: (Each undeclared identifier is reported only once app_addon_sql_mysql.c:164: error: for each function it appears in.) make: *** [app_addon_sql_mysql.o] Error 1 Any ideas? Same code, so I don't think there's a problem with the code itself. It has to be some lib issue or weirdness. Anyone run into this before? N. ___ --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] Issues compiling addons on Fedora Core 3
On Sat, Aug 12, 2006 at 01:48:40PM -0400, sip wrote: I have zero problem compiling the addons 1.23 on FC4 and RH4, but for some reason, when I try to compile them on FC3, I get this: cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:164:36: macro AST_LIST_REMOVE requires 4 arguments, but only 3 given app_addon_sql_mysql.c: In function `del_identifier': app_addon_sql_mysql.c:164: error: `AST_LIST_REMOVE' undeclared (first use in this function) app_addon_sql_mysql.c:164: error: (Each undeclared identifier is reported only once app_addon_sql_mysql.c:164: error: for each function it appears in.) make: *** [app_addon_sql_mysql.o] Error 1 Where is the asterisk source directory? What version of asterisk do you have? Any ideas? Same code, so I don't think there's a problem with the code itself. It has to be some lib issue or weirdness. Anyone run into this before? N. ___ --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 -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.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] Issues compiling addons on Fedora Core 3
Asterisk source dir is one level up from this. Basically (and this is the same way it compiles on our other servers): /home/user -/home/user/asterisk-1.2.10 -/home/user/asterisk-addons-1.2.23 On Tue, 15 Aug 2006 15:59:04 +0300, Tzafrir Cohen wrote On Sat, Aug 12, 2006 at 01:48:40PM -0400, sip wrote: I have zero problem compiling the addons 1.23 on FC4 and RH4, but for some reason, when I try to compile them on FC3, I get this: cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:164:36: macro AST_LIST_REMOVE requires 4 arguments, but only 3 given app_addon_sql_mysql.c: In function `del_identifier': app_addon_sql_mysql.c:164: error: `AST_LIST_REMOVE' undeclared (first use in this function) app_addon_sql_mysql.c:164: error: (Each undeclared identifier is reported only once app_addon_sql_mysql.c:164: error: for each function it appears in.) make: *** [app_addon_sql_mysql.o] Error 1 Where is the asterisk source directory? What version of asterisk do you have? Any ideas? Same code, so I don't think there's a problem with the code itself. It has to be some lib issue or weirdness. Anyone run into this before? N. ___ --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 -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.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
[asterisk-users] extensions.ael - calling an exten from a macro
Hi,
I'm trying to call an exten from inside extensions.ael, as below, ddi calls
ael and then ael needs to call the extensions.conf (8000 exten) for the call
queue.
Is this possible? Or is there an easier way to combine the exten 8000 to the
ael?
Thanks,
Dean.
ddi.conf
exten = _441234567890,1,Macro(queueexten-ael,8000)
extension.ael
macro queueexten-ael( ext ) {
if (${CALLERID(num):0:2} = 44) {
Set(CALLERID(num)=0${CALLERID(num):2});
} else
Set(CALLERID(num)=00${CALLERID(num)});
Dial(8000); --- don't know this bit!
}
extensions.conf
[8000]
; Forecourt Services Call Queue
include = daytime|8:00-18:00|Mon-Fri|*|*
include = night|18:00-8:00|Mon-Fri|*|*
include = night|*|Sat-Sun|*|*
[daytime]
exten = 8000,1,Answer
exten = 8000,2,Set(CALLERID(NAME)=Forcourt Services)
exten = 8000,3,Queue(fservices1800)
[night]
exten = 8000,1,Playback(/var/lib/asterisk/sounds/fsdeskclosed)
exten = 8000,2,Wait(2)
exten = 8000,3,Hangup()
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[asterisk-users] FAX questions
Hello, I have an incoming PRI (T405P card) and a TDM 2400P with 6 modules connected to a channel bank with 50 pins cable. There are places for 16 phones, 4 faxes and 4 GSM gateway. Outgoing call are perfect but I will need some debounce tuning because sometimes line ring back. But it's another story. I use 2 faxes (Zap/7 and Zap/5) and they collected in group. Zap/7 is my primary fax and Zap/5 is my secondary fax. I write a simple macro for incoming faxes: [macro-fax-incoming] exten = s,1,Dial(Zap/G3) exten = s,2,Hangup exten = s,102,Hangup I think there is enough room for improving this macro. :-) G3 because I need to go down in channel numbers (7-5). Do I need Answer() or this direct connection with Dial() is enough? If I use Answer() the caller will pay even if both faxes are busy. Do I need send busy tone and give a Busy() command if both channels are busy or is it automatically happen? My problem is I get a lot of 'NO ANSWER' call in CDR when both faxes are ready. How I store DIALSTATUS in CDR record? Another question. With latest version of asterisk softwares am I able using rxfax? I had read some remarks about incompatibility between TDM card and rxfax. Is it still exist? bye, Zsolt ___ --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 load testing
Nitin: Some generalized specs: A voice call takes aprox. 30MHZ of CPU. In your spec, a dual 240 (1.4Ghz) may take up to: 1400/30=46 calls x 2 = 92 calls Im just talking G711 here. I have not taken into account if you're going to use voicemail, AGI, etc,etc,. Just plain calls. I also have not taken into account how many phones can register to this machine. Personally, I make calls, not registrations, so it is useless to me to know that a billion phones can register to a given asterisk machine but only 100 can make calls. So, my personal point of view is that your machine can do 92 calls (SIP TO ZAP) at full g711 quality with at least 4 times the registrations (that means about 400 phones can register). However, due to the CPU structure of Opterons, that number may be a little high. As Martin said, look the archives. There are gallizions of configurations that can help you, or, use/rent products like ABACUS or the asterisk load tester. And about howe much internet bandwidth a codec requieres, well, look for the codec size/payload and add a few kilobits of IP overhead. Example: G711 is 64 kilobits per second, a conservative figure will be to add 16 kilobits of overhead so the total size of a g711 transmission will be (64+16) 80 Kilobits per second per leg. When you see the term per leg it means this: SIP user/g711-80kbps(first leg)-Asterisk80kbps(second leg)-destination That means each side of the conversation will take 80Kbps of bandwidth. Hope it helps, feel free to ask again and welcome to the list. Cheeers, On 8/14/06, Nitin Gupta [EMAIL PROTECTED] wrote: Hi, did anyone try do load-testing on asterisk, for sip channel calls? I want to have a rough estimate about - how many calls, an asterisk server, running on say dual 240 opteron with 1 GB memory, can handle? Also how much internet bandwidth does a typical call requires? I heard around 20Kbps with typical codecs, is that right? Thanks in advance, Nitin ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Hangup Problem with PSTN and ISDN
Hi, sipphone -- Asterisk PBX -- PSTN -- Cell Phone sipphone sets up a call to Cell Phone. When Cell Phone hangs up, it takes about 1 minute for Asterisk to show the Hangup Zap 1/1 message, after which sipphone hangs up. During the time before Asterisk shows the Hangup message, Busy Tone can be heard at sipphone. Does anyone know why Asterisk took 1 minute to hangup ? Am I right to say that Disconnect Supervision is enabled in PSTN ? Is the Busy Tone generated by Asterisk ? If that is so, then Asterisk must have known that the line is hung up. I conducted another experiment. sipphone -- Asterisk PBX -- ISDN -- PSTN -- Cell Phone sipphone sets up a call to Cell Phone. When Cell Phone hangs up, Asterisk does not hangup at all. From the ISDN messages, it shows that Asterisk receives the Disconnect message and seem to be disconnected from ISDN. However, there isn't any Hangup message shown. There isn't any tone at sipphone when Cell Phone hangs up. Is the ISDN Hangup problem related to Disconnect Supervision ? Thank you. Regards, Kwang Mien ___ --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] 1 way audio. Dual NIC's.
That did the trick. Thanks for the tip. Interesting though. Although technically it is behind a NAT, it is also connecting with the server who is also behind the NAT, I figured that in the eyes of the server... it would need NAT=no because neither device is connecting to it *through* the NAT. Whatever... thanks a million. bp On 8/15/06, Earl Terwilliger [EMAIL PROTECTED] wrote: how aboutnat=yesqualify=yescanreinvite=noaccording to: http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions ___ --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] Zap difficulties
That did help. But can you help me understand why this is needed? I did not notice any of the other issues you mentioned but I do notice that it takes an unusually long time to hang up the channel when it is done with the call. It almost seems like the signaling is not right. I was discussing this issue with someone offline and from what I understand, the POTS lines are on loopstart. If that is true why do we use koolstart on the zaptel channel? Just as an experiment I did change the signaling to loopstart but that did not help either. The biggest issue is that I am in an area where just about all of the business are using POTS lines exclusively, and adding a pause to all of these just seems like a hack to me rather than fixing an issue. I'm not saying this is not my misunderstanding, because it may well be, but I am just looking for the exact answer. Thanks Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Tuesday, August 15, 2006 12:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap difficulties Curt Shaffer wrote: I am having a weird issue with my zap channel (Digium TDM01B). Randomly it appears that the POTS line is not seeing all of the digits passed. We have to dial a 1 and the area code to call most numbers here, and we get the error that we need to dial a 1 and the area code when dialing this number even though we are dialing it. Also when I dial 8xx numbers it never works (same error). I do have all of those set up as allowed and routing properly from the dial plan and I can test that by switching to a VoIP termination and the calls go through without a hitch. I can also dial these numbers fine if I hook a POTS phone directly to the cable that connects to the Digium card. Asterisk looks as if it is passing the digits, (ZAP/g0/18003569377|120|r) for example. Dial(ZAP/g0/w18003569377|120) This will put a .5 second wait before dialing to allow the telco equipment to get ready to receive DTMF. Have you noticed other issues like, even when calling busy numbers, you hear a ringing tone for about 5.5 seconds before you hear a busy tone? That's because you are using the r option to Dial. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] ARI
In ARI, after logging in to check voicemail, I am getting the following error:"Voicemail recording(s) was not found.On settings page, change voicemail audio format. It is currently set to .gsm"I used to play the voicemail to check voicemail. It used to work, suddenly i get the above error. I tried to change the audio format to other options in the pull down, didn't help. I am clueless now.Thanks,-balu ramanRyder Brook PediatricsP.O.Box 608Morrisville, VT 05661 __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] Manager Interface API's
Can anyone recommend the best Manager Interface API, putting language preferences aside? The python and perl ones have bupkiss documentation. I can't understand why anyone would even write an api and make it publically available without documenting it. 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
Re: [asterisk-users] Manager Interface API's
Douglas Garstang wrote: Can anyone recommend the best Manager Interface API, putting language preferences aside? Asterisk-Java of course ;) http://asterisk-java.org/latest for the stable release and http://asterisk-java.org/0.3-SNAPSHOT for the dev snapshot. Includes a short tutorial and javadoc for everything else. The python and perl ones have bupkiss documentation. I can't understand why anyone would even write an api and make it publically available without documenting it. 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 -- reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: [EMAIL PROTECTED] signature.asc Description: OpenPGP digital 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] Manager Interface API's
Probably cause it is someone like most of us sitting at home doing it...releasing it for free...so why would we write pages of documentation for it? If it's open source and it's free...Then offer them some money to make documentation for it hehe... - Original Message - From: Douglas Garstang To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, August 15, 2006 11:05 AM Subject: [asterisk-users] Manager Interface API's Can anyone recommend the best Manager Interface API, putting language preferences aside? The python and perl ones have bupkiss documentation. I can't understand why anyone would even write an api and make it publically available without documenting it. Doug. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message.Checked by AVG Free Edition.Version: 7.1.405 / Virus Database: 268.10.10/419 - Release Date: 8/15/2006 ___ --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] FAX questions
Hi, Another question. With latest version of asterisk softwares am I ableusing rxfax? I had read some remarks about incompatibility between TDMcard and rxfax. Is it still exist? I've been using rx for fax reception with TE110P as well as X100P (this only for tests and learning) with very success. As far as i know what could be a problem is that SpanDSP doesn't implements ECM (error correction mode) For Fax reception, only with X100P i've had to play with rxgains, nothing else. I've had some problems only for tx fax lots of errors transmiting faxs, but i think that could be because my * is behind a legacy pbx and i could be facing time sinchronization problems. bye,Zsolt ___ --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] Zap difficulties
It's normal to have to wait (under a second in your case) for a dial tone from the phone company when seizing a line. If you were placing a call on a phone directly connected to the phone company, the time it takes to physically pick up the phone and move your hand to the dial normally takes at least a half a second, giving the CO time to start the dial tone and prepare to receive the dialed digits. In the old days, one actually had to listen for the dial-tone before dialing, as the phone company equipment would not necessarily be ready to receive your digits in 1-2 seconds. With modern electronic switches, though, a constant delay of 0.5s - 1.0s should be fine. -Rusty On 8/15/06, Curt Shaffer [EMAIL PROTECTED] wrote: That did help. But can you help me understand why this is needed? I did not notice any of the other issues you mentioned but I do notice that it takes an unusually long time to hang up the channel when it is done with the call. It almost seems like the signaling is not right. I was discussing this issue with someone offline and from what I understand, the POTS lines are on loopstart. If that is true why do we use koolstart on the zaptel channel? Just as an experiment I did change the signaling to loopstart but that did not help either. The biggest issue is that I am in an area where just about all of the business are using POTS lines exclusively, and adding a pause to all of these just seems like a hack to me rather than fixing an issue. I'm not saying this is not my misunderstanding, because it may well be, but I am just looking for the exact answer. Thanks Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Tuesday, August 15, 2006 12:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap difficulties Curt Shaffer wrote: I am having a weird issue with my zap channel (Digium TDM01B). Randomly it appears that the POTS line is not seeing all of the digits passed. We have to dial a 1 and the area code to call most numbers here, and we get the error that we need to dial a 1 and the area code when dialing this number even though we are dialing it. Also when I dial 8xx numbers it never works (same error). I do have all of those set up as allowed and routing properly from the dial plan and I can test that by switching to a VoIP termination and the calls go through without a hitch. I can also dial these numbers fine if I hook a POTS phone directly to the cable that connects to the Digium card. Asterisk looks as if it is passing the digits, (ZAP/g0/18003569377|120|r) for example. Dial(ZAP/g0/w18003569377|120) This will put a .5 second wait before dialing to allow the telco equipment to get ready to receive DTMF. Have you noticed other issues like, even when calling busy numbers, you hear a ringing tone for about 5.5 seconds before you hear a busy tone? That's because you are using the r option to Dial. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Problems with incoming authentication
I'm far more interested in finding out a solution to my problem, but:1 When everything is set up correctly, they're just fine.2 Yes, except for the problem I originally posted to start this thread.3 I don't know, it all depends on you. 4 I don't know, VoicePulse is the only think I've tested with so far.On 8/15/06, Crazy Boy [EMAIL PROTECTED] wrote:Hi,I am Chandra from India. We have Installed Asterisk in our organization. We want to buy a VoIP plan to make calls to US. I have some doubts. Please clarify. 1) How is the Voicepulse service?2) Is Voicepulse working fine with Asterisk?3) Can I configure Voicepulse easily with Asterisk?4) From Teliax and Voicepulse, Which is offering better service?Looking forward to your response. Thank you. Regards,Chandra.David Freeman [EMAIL PROTECTED] wrote: Sometimes I can receive a call to my DID, but sometimes it just rings and rings and I see these messages in the full log:[Aug 14 23:32:12] DEBUG[3556] res_crypto.c: Key failed verification: voicepulse20060419 [Aug 14 23:32:12] NOTICE[3556] chan_iax2.c: Host 64.61.93.87 failed to authenticate as voicepulse[Aug 14 23:32:13] DEBUG[3550] res_crypto.c: Key failed verification: voicepulse20060419[Aug 14 23:32:13] NOTICE[3550] chan_iax2.c: Host 64.61.93.90 failed to authenticate as voicepulseUsually, when this happens, I can immediately re-dial the DID and * receives the call and sends it on the dialplan.I've tried configuring my IAX2 user details to no use rsa and no key, but the problem persists. I've tried to delete my IAX2 trunks to just use the SIP ones and I get the same problem...in fact, if I only have SIP trunks, I can't receive any calls to the DID.I'm using Asterisk SVN-trunk-r39753M currently, updated today. Any help would be appreciated, I'm running out of ideas. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Stay in the know. Pulse on the new Yahoo.com. Check it out. Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail Beta. ___--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
RE: [asterisk-users] Manager Interface API's
Well, I don't know about you, but if I have to read the source code to work out how it works, I'm going to go and look at someone elses, that may have some BASIC documentation and examples. -Original Message-From: Don [mailto:[EMAIL PROTECTED]Sent: Tuesday, August 15, 2006 9:09 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Manager Interface API's Probably cause it is someone like most of us sitting at home doing it...releasing it for free...so why would we write pages of documentation for it? If it's open source and it's free...Then offer them some money to make documentation for it hehe... - Original Message - From: Douglas Garstang To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, August 15, 2006 11:05 AM Subject: [asterisk-users] Manager Interface API's Can anyone recommend the best Manager Interface API, putting language preferences aside? The python and perl ones have bupkiss documentation. I can't understand why anyone would even write an api and make it publically available without documenting it. Doug. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message.Checked by AVG Free Edition.Version: 7.1.405 / Virus Database: 268.10.10/419 - Release Date: 8/15/2006 ___ --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] Sending SIP 183 Session Progressing
Dinesh, I suspect your problem is with the softphone implementation... I have an Asterisk PBX setup with ISDN (chan_capi) and use Cisco 7960 phones with Cisci SIP 7.5 firmware and get to watch the various SIP messages in/out on the phone. Depending on the phone numbers I dial (and the signalling back from the ISDN exchange) I get 100 - 183 - 180 or 100 - 180 In both cases the Cisco plays our ringing on receipt of the 180. Occasionally calls which go from 100 - 180 without going via the 183 result in the Cisco ringing and combined rining genrated by the telephone exchange which is weird but ok. I have also encountered (rarely) ISDN number which, when dialled from 100 - 183 - Connected without a ringing phase - these call result in silence at the Cisco phone followed by connected audio (from the far end) - which is to be expected. Mike - Original Message - From: Dinesh Nair [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, August 15, 2006 7:18 AM Subject: [asterisk-users] Sending SIP 183 Session Progressing i'm not sure if this is a -users or a -dev question, but am sending it here anyways. discussion could move to -dev if chan_sip.c code needs to be amended/explained. first up, all this on asterisk 1.2.10 on freebsd 6.1. here's the beef: from a particular sip softphone we're playing with, we notice that calls to another SIP phone (same LAN) result in the /lack/ of a ringing tone on the softphone. however, calls from the same softphone to a PSTN/Mobile number (through a TE405P) result in proper behaviour on the softphone with a ringing tone. an ethereal trace of both types of calls results in only one difference. for calls to the PSTN/Mobile thru libpri/chan_zap, asterisk returns a SIP 183 Session Progress[1] packet in between the 100 Trying and 180 Ringing, while for calls from the softphone to another SIP phone it's 100 Trying followed immediately by 180 Ringing. so my question is, is the softphone behaving correctly in not playing a ringing tone to the user without the 183 packet inspite of the 180 Ringing packet being received ? alternatively, since we aren't able to change the softphone, will i break anything big if i force asterisk to send the 183 packet immediately after sending the 100 Trying packet in sip_indicate() ? alternatively, in reading the RFCs, i came across RFC3398 which speficies mappings between ISDN Cause Codes and SIP responses. has this mapping been implemented in asterisk at the moment, either in 1.2 or the upcoming 1.4 ? [1] the 183 Session Progress packet is triggered by the receipt of a PRI PROGRESS indicator from libpri, which gets translated to a AST_CONTROL_PROGRESS and thence a 183 Session Progress to SIP. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.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 ___ --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] Zap difficulties
Also, unfortunately, Asterisk does NOT listen for dialtone before dialing, so these problems will continue until someone sees fir to fix it. As an aside, for those who pulse dial, rather than DTMF, the w will not work, as it only works in DTMF John Novack Rusty Dekema wrote: It's normal to have to wait (under a second in your case) for a dial tone from the phone company when seizing a line. If you were placing a call on a phone directly connected to the phone company, the time it takes to physically pick up the phone and move your hand to the dial normally takes at least a half a second, giving the CO time to start the dial tone and prepare to receive the dialed digits. In the old days, one actually had to listen for the dial-tone before dialing, as the phone company equipment would not necessarily be ready to receive your digits in 1-2 seconds. With modern electronic switches, though, a constant delay of 0.5s - 1.0s should be fine. -Rusty On 8/15/06, Curt Shaffer [EMAIL PROTECTED] wrote: That did help. But can you help me understand why this is needed? I did not notice any of the other issues you mentioned but I do notice that it takes an unusually long time to hang up the channel when it is done with the call. It almost seems like the signaling is not right. I was discussing this issue with someone offline and from what I understand, the POTS lines are on loopstart. If that is true why do we use koolstart on the zaptel channel? Just as an experiment I did change the signaling to loopstart but that did not help either. The biggest issue is that I am in an area where just about all of the business are using POTS lines exclusively, and adding a pause to all of these just seems like a hack to me rather than fixing an issue. I'm not saying this is not my misunderstanding, because it may well be, but I am just looking for the exact answer. Thanks Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Tuesday, August 15, 2006 12:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap difficulties Curt Shaffer wrote: I am having a weird issue with my zap channel (Digium TDM01B). Randomly it appears that the POTS line is not seeing all of the digits passed. We have to dial a 1 and the area code to call most numbers here, and we get the error that we need to dial a 1 and the area code when dialing this number even though we are dialing it. Also when I dial 8xx numbers it never works (same error). I do have all of those set up as allowed and routing properly from the dial plan and I can test that by switching to a VoIP termination and the calls go through without a hitch. I can also dial these numbers fine if I hook a POTS phone directly to the cable that connects to the Digium card. Asterisk looks as if it is passing the digits, (ZAP/g0/18003569377|120|r) for example. Dial(ZAP/g0/w18003569377|120) This will put a .5 second wait before dialing to allow the telco equipment to get ready to receive DTMF. Have you noticed other issues like, even when calling busy numbers, you hear a ringing tone for about 5.5 seconds before you hear a busy tone? That's because you are using the r option to Dial. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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
[asterisk-users] Can budgetone 101 display name part of cid?
Hi, I've been trying to get this to work, but I'm not havinf much luck. So is it even possible to get a budgetone 101 to show the text bit of the callerid=some text 1234567890? It shows the number just fine (after explicitly setting it with Set(CALLERID(all)=some text 123-123-1234). It can show text there, because I've seen messages like Tr and Unkn (I think) appearing in other situations. A softphone like sjphone shows the textbit, even without the Set, just a callerid entry in sip.conf for the phone is enough. TIA! Regards, Guus ___ --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] Can budgetone 101 display name part of cid?
Try without the Set(CALLERID(all)=some text 123-123-1234 Julian Guus Houtzager wrote: Hi, I've been trying to get this to work, but I'm not havinf much luck. So is it even possible to get a budgetone 101 to show the text bit of the callerid=some text 1234567890? It shows the number just fine (after explicitly setting it with Set(CALLERID(all)=some text 123-123-1234). It can show text there, because I've seen messages like Tr and Unkn (I think) appearing in other situations. A softphone like sjphone shows the textbit, even without the Set, just a callerid entry in sip.conf for the phone is enough. TIA! Regards, Guus ___ --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] Can budgetone 101 display name part of cid?
On Tuesday 15 August 2006 17:53, Julian Lyndon-Smith wrote: Try without the Set(CALLERID(all)=some text 123-123-1234 Tried that, no effect, still shows only the number part, without the '-'s though. I find it weird I should have to Set it explicitly, why does a softphone like sjphone show what's in the callerid field of the phone in sip.conf (as I think it should be) and the budgetone does not? Implementation difference in sip protocol? Julian Regards, Guus ___ --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] Page Groups
I have a company that I am going to be moving away from a legacy PBX to Asterisk. They use page zones pretty heavy and I would like to keep that functionality. Basically when someone is not at their desk the receptionist pages all of the phones, telling them there is a call. Does anyone out there know of the best phones to do this with and if it is really even possible. I see that intercom is not supported and paging appears to be minimally supported. Thanks Curt ___ --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] Intel D945G chipset
Any problem running Asterisk w/ Digium hardware with motherboards using that chipset (for example the D945GPM) ftp://download.intel.com/design/motherbd/pm/D3610601US.pdf I was thinking of running a TE212P. Bill ___ --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] Can budgetone 101 display name part of cid?
Guus Houtzager wrote: Hi, I've been trying to get this to work, but I'm not havinf much luck. So is it even possible to get a budgetone 101 to show the text bit of the I'm not sure about the 101, but the Budgetone 100 is only capable of numeric data. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager Interface API's
Douglas. Please take this as a constructive comment. I have followed your questions in asterisk-dev and users lists, and you always seem to make non constructive comments about the people giving code/work for Free. And you focus in the negative part, never giving importance to the positive things about it. If you dont like something, then change it yourself, they are not providing a payed service. The source is available AS-IS if you want it, and if you like it, take it; If you dont, just ignore it, try to not make peyorative comments. Regards On 8/15/06, Douglas Garstang [EMAIL PROTECTED] wrote: Well, I don't know about you, but if I have to read the source code to work out how it works, I'm going to go and look at someone elses, that may have some BASIC documentation and examples. -Original Message- From: Don [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 15, 2006 9:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Manager Interface API's Probably cause it is someone like most of us sitting at home doing it...releasing it for free...so why would we write pages of documentation for it? If it's open source and it's free...Then offer them some money to make documentation for it hehe... - Original Message - From: Douglas Garstang To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, August 15, 2006 11:05 AM Subject: [asterisk-users] Manager Interface API's Can anyone recommend the best Manager Interface API, putting language preferences aside? The python and perl ones have bupkiss documentation. I can't understand why anyone would even write an api and make it publically available without documenting it. 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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.10.10/419 - Release Date: 8/15/2006 ___ --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] Can budgetone 101 display name part of cid?
The Budgetone only supports a 12-digit caller ID LCDOn 8/15/06, Guus Houtzager [EMAIL PROTECTED] wrote: On Tuesday 15 August 2006 17:53, Julian Lyndon-Smith wrote: Try without the Set(CALLERID(all)=some text 123-123-1234Tried that, no effect, still shows only the number part, without the '-'sthough.I find it weird I should have to Set it explicitly, why does a softphone like sjphone show what's in the callerid field of the phone in sip.conf (as Ithink it should be) and the budgetone does not? Implementation difference insip protocol? JulianRegards,Guus ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephonywww.baldwintechsolutions.comPhone: 518-631-2855 x205Fax: 518-631-2856 ___ --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] Page Groups
Any phone that supports Auto answer can do this, among those phones: Cisco 796x, Polycom 3xx,430,50x,60x, SPA9xx. The SPA9xx (which support auto answer) will even support it while you are on the phone, it will however put the current conversation on hold for the duration of the page. On 8/15/06, Curt Shaffer [EMAIL PROTECTED] wrote: I have a company that I am going to be moving away from a legacy PBX to Asterisk. They use page zones pretty heavy and I would like to keep that functionality. Basically when someone is not at their desk the receptionist pages all of the phones, telling them there is a call. Does anyone out there know of the best phones to do this with and if it is really even possible. I see that intercom is not supported and paging appears to be minimally supported. Thanks Curt ___ --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] Manager Interface API's
I, for one, didn't take his comment as anything other than constructive Lack of documentation is an issue, open source or not. It is an unfortunate situation that many very smart coders understand what they have created, but are unwilling or unable to supply enough information for many others to make effective use of their creation How many have struggled through the years with uncommented or poorly commented code when the original creator is off to greener pastures? JMO John Novack Moises Silva wrote: Douglas. Please take this as a constructive comment. I have followed your questions in asterisk-dev and users lists, and you always seem to make non constructive comments about the people giving code/work for Free. And you focus in the negative part, never giving importance to the positive things about it. If you dont like something, then change it yourself, they are not providing a payed service. The source is available AS-IS if you want it, and if you like it, take it; If you dont, just ignore it, try to not make peyorative comments. Regards On 8/15/06, Douglas Garstang [EMAIL PROTECTED] wrote: Well, I don't know about you, but if I have to read the source code to work out how it works, I'm going to go and look at someone elses, that may have some BASIC documentation and examples. -Original Message- From: Don [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 15, 2006 9:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Manager Interface API's Probably cause it is someone like most of us sitting at home doing it...releasing it for free...so why would we write pages of documentation for it? If it's open source and it's free...Then offer them some money to make documentation for it hehe... - Original Message - From: Douglas Garstang To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, August 15, 2006 11:05 AM Subject: [asterisk-users] Manager Interface API's Can anyone recommend the best Manager Interface API, putting language preferences aside? The python and perl ones have bupkiss documentation. I can't understand why anyone would even write an api and make it publically available without documenting it. 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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.10.10/419 - Release Date: 8/15/2006 ___ --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] Page Groups
For
intercom, do you mean placing a call that is automatically answered by the
called party?
If so,
the following works for legacy phones connected via a Citel Handset Gateway,
amongst others:
exten
= _*803X.,1,Macro(user-callerid)exten =
_*803X.,2,SetVar(_ALERT_INFO=info=alert-autoanswer)
exten
= _*803X.,3,SIPAddHeader(Answer-Mode: Auto) exten =
_*803X.,4,Dial(SIP/${EXTEN:4})
(so
you dial *803 and then the extension number you want to
target)
Similar techniques can be used for page.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Curt
ShafferSent: 15 August 2006 17:16To: 'Asterisk Users
Mailing List - Non-Commercial Discussion'Subject: [asterisk-users]
Page Groups
I have a company that I am going
to be moving away from a legacy PBX to Asterisk. They use page zones pretty
heavy and I would like to keep that functionality. Basically when someone is
not at their desk the receptionist pages all of the phones, telling them there
is a call. Does anyone out there know of the best phones to do this with and
if it is really even possible. I see that intercom is not supported and paging
appears to be minimally supported.
Thanks
Curt
___
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Re: [asterisk-users] Can budgetone 101 display name part of cid?
Doug,That is correct you can only display the number on the BudgetTone 101, 102, and 200.If you wish to display the name as well, you will need to upgrade to the GXP-2000 phone. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Aug 15, 2006, at 11:19 AM, Doug Lytle wrote:Guus Houtzager wrote: Hi,I've been trying to get this to work, but I'm not havinf much luck. So is it even possible to get a budgetone 101 to show the text bit of the I'm not sure about the 101, but the Budgetone 100 is only capable of numeric data.Doug-- Ben Franklin quote:"Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."___--Bandwidth and Colocation provided by 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
Re: [asterisk-users] Ringing after answered on zaptel
Try setting: progressinband=no in your sip.conf -Brodie On Monday 14 August 2006 10:20 pm, Don Fanning wrote: Greetings List, I'm having a strange problem with my X100p card still ringing after the call is connected. Any idea on how to solve this? Using latest asterisk (not svn) along with latest zaptel driver. Thanks, Don ___ --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] Can budgetone 101 display name part of cid?
Jessee J Holmes wrote: Doug, That is correct you can only display the number on the BudgetTone 101, 102, and 200. If you wish to display the name as well, you will need to upgrade to the GXP-2000 phone. I'm not, Guus is. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager Interface API's
John Novack wrote: I, for one, didn't take his comment as anything other than constructive Lack of documentation is an issue, open source or not. To make this thread even more constructive: What kind of documentation do you expect from a Manager API package? What features do you expect? - A plain wrapper for the Actions, Responses and Events? - An abstracted view on Asterisk's concept like channels, extensions, queues and so on? And last not least: Would a language independant specification help? Something like: There is a channel concept (object) with the properties id, name, caller id, ... and the operations hangup, redirect, ... =Stefan -- reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: [EMAIL PROTECTED] Jabber: [EMAIL PROTECTED] signature.asc Description: OpenPGP digital 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] Manager Interface API's
Douglas Garstang wrote: Can anyone recommend the best Manager Interface API, putting language preferences aside? The python and perl ones have bupkiss documentation. I can't understand why anyone would even write an api and make it publically available without documenting it. Have you taken your be nice on the lists pill today? The most likely explanation is that people have written these interfaces primarily for their own use, and when they decided to share with others, only had/made time to minimally document them. Do you understand that? You've got me doubting you can't understand such things, so I wonder why you *say* you don't understand. Unless you enjoy being a troll. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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 asterisk over Linksys VPN
Has anybody tried using a VPN and around 10 phones behind the tunnel to connect to an asterisk server using Linksys VPN routers? Like this one: http://www.linksys.com/servlet/Satellite?c=L_Product_C2childpagename=US%2FLayoutcid=1115416832495pagename=Linksys%2FCommon%2FVisitorWrapper ___ --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] Manager Interface API's
John Novack wrote: I, for one, didn't take his comment as anything other than constructive Lack of documentation is an issue, open source or not. It is an unfortunate situation that many very smart coders understand what they have created, but are unwilling or unable to supply enough information for many others to make effective use of their creation How many have struggled through the years with uncommented or poorly commented code when the original creator is off to greener pastures? I have struggled like that on a great number of occasions, and know perfectly what you are describing. But I don't think it's fair to blame people in the Open Source community for not doing pro-grade documentation. They give away what they write; if it's useful, all good. If not, then buy a commercial product, or move to another OS product that has better documentation. Especially in this case, where the overwhelming likelihood is that the programmers wrote the APIs primarily for their own use, I don't think it's fair to be casting Garstangian aspersions. Those APIs aren't big public projects, but rather labors of love that don't have the kind of support staff to handle a robust public face. MO. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] 1.2.10 - g726 Issues
I have hard that 1.2.10 has issues with voice quality through g726. Can anyone provide any feedback or point me in the right direction about the current status of this problem? Thanks, Cullin ___ --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] Manager Interface API's
On 8/15/06, John Novack [EMAIL PROTECTED] wrote: I, for one, didn't take his comment as anything other than constructive Yes, I agree is possible just lack of niceness. But may be you dont have idea of the bunch of peyorative comments about IAX2 protocol he made in asterisk-dev list without doing a good proposal. Lack of documentation is an issue, open source or not. Is an issue, but at least for me, it seems that just complaining, without a good proposal, is just worse. And in fact is not an issue for those who can really understand the code. So, if its an issue for you, pay someone to do it, or doit yourself. The difference between open source, and the others, is that for open source usually you dont pay, for the commercial software, of course i would be expecting documentation. It is an unfortunate situation that many very smart coders understand what they have created, but are unwilling or unable to supply enough information for many others to make effective use of their creation That is because THEY DONT CARE, they are putting you one or more steps forward in the *right* direction, is not they responsability to provide documentation, as I said, they provide code AS IS, in the hope that will be usefull to someone with enough skills to understand the code. And also, hopefully, some one else will create the documentation, or even the developer, when he/she has the free time. But always remember, they are giving for FREE their time. So the better we can do is ask kindly for documentation. How many have struggled through the years with uncommented or poorly commented code when the original creator is off to greener pastures? Then, why dont you make it better? Open Source should be a community effort, not just developers efforts. I agree that the best person to document what the code does is the developer him/her self, but, again, is not responsible for doing so, since no one of us are paying for it. -- moy 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] Unable to receive Incoming calls to my DID. Please tell me the solution
Hi My suggestion is run Ngrep is the best tool to see what is happening is the DID calling from PSTN, the call is landing in the server or not So better option is that to check with Ngrep DID will give you details Ram On 8/15/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi,Thank you very much for your patience to give solutions for me. Today is holiday for us because of our Independance day. Tomorrow I will do and check as suggested by you and let you know. Once again, Thank you. Regards,Chandra. Lacy Moore - Aspendora [EMAIL PROTECTED] wrote: I struggled with one provider for a long time until finally realizing my username on their site was not my username that I was supposed to be using in sip configuration. Make sure you are using the right username and password. However, it would seem that you would not be able to make an outgoing call using the wrong username/password combination. One thing I have not seen in your posts is your firewall information. Your firewall may be setup to allow outgoing connections, but not incoming. I would not depend on info from a provider. You may very well be registering with them, but your firewall may be blocking the incoming call. If you think you have no firewall, check again. IPTABLES might have loaded itself and it may be blocking. Try: service iptables stop and then try the incoming call again. I've been burned twice due to this. Something has changed in the way I configure my linux boxes, and for some reason iptables is starting. On 8/14/06, Rich Adamson [EMAIL PROTECTED] wrote: Crazy Boy wrote: Hi, Thank you for your response. As you said, I executed the command sip show registry. But, its not showing anything. Teliax people are also telling that my Asterisk server doesn't register with Teliax. So, the final conclusion is My Asterisk server doesn't register with Teliax. Here I am giving my configuration files. Now, What I have to do to register my Asterisk server with Teliax? Please tell me. SIP.CONF contents: [general] register = xyz.abc:[EMAIL PROTECTED] [authentication] auth = xyz.abc:[EMAIL PROTECTED] Double check the above two statements to ensure the userid and passwordare exactly those provided to you by teliax. There is nothing else inyour config that impacts the register statement with the exception of nat'ing.It would appear from your other config statements that asterisk might belocated behind a firewall or nat box. If so, read the documentation onthat, and look at the asterisk/configs/sip.conf.sample file. Specifically the section on NAT SUPPORT.You might also want to read more about using the diagnostic toolsavailable to you within asterisk. Setting verbose and/or debug to a highnumber and copy/paste the CLI output associated with the problem. Or, start using the CLI with something like:asterisk -rvv [teliax-incoming] exten = 3031234567, 1, Answer() exten = 3031234567, 2, Dial(SIP/105,15) The above has nothing to do with registering with teliax, but you do notwant to answer a call before ringing the sip phone. Take thatstatement out of there. When the sip phone answers an incoming call, asterisk will automatically send the answer to teliax.___--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-- Lacy Moore Aspendora, Inc. ___ --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 Do you Yahoo!?Next-gen email? Have it all with the all-new Yahoo! Mail Beta. ___--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
Re: [asterisk-users] Ringing after answered on zaptel
That's kind of useless since progressinband only applies to digital interfaces. Try callprogress=no Brodie Macleod wrote: Try setting: progressinband=no in your sip.conf -Brodie On Monday 14 August 2006 10:20 pm, Don Fanning wrote: Greetings List, I'm having a strange problem with my X100p card still ringing after the call is connected. Any idea on how to solve this? Using latest asterisk (not svn) along with latest zaptel driver. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Page Groups
For paging, and I have not done this yet,
you would probably have to invite all the phones to a conference with the
auto-answer
The below works great for intercom though .
Polycom which I have used
exten = _*7XXX,1,SetVar(ALERT_INFO=Ring
Answer)
exten = _*7XXX,2,Dial.blah
Bill
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Langstaff
Sent: Tuesday, August 15, 2006
12:46 PM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Page
Groups
For intercom, do you mean placing a call
that is automatically answered by the called party?
If so, the following works for legacy
phones connected via a Citel Handset Gateway, amongst others:
exten = _*803X.,1,Macro(user-callerid)
exten = _*803X.,2,SetVar(_ALERT_INFO=info=alert-autoanswer)
exten =
_*803X.,3,SIPAddHeader(Answer-Mode: Auto)
exten = _*803X.,4,Dial(SIP/${EXTEN:4})
(so you dial *803 and then the extension
number you want to target)
Similar techniques can be used for page.
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Curt Shaffer
Sent: 15 August 2006 17:16
To: 'Asterisk
Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Page
Groups
I have a company that I am going to be moving away from a
legacy PBX to Asterisk. They use page zones pretty heavy and I would like to
keep that functionality. Basically when someone is not at their desk the
receptionist pages all of the phones, telling them there is a call. Does anyone
out there know of the best phones to do this with and if it is really even
possible. I see that intercom is not supported and paging appears to be
minimally supported.
Thanks
Curt
___
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Re: [asterisk-users] Manager Interface API's
Some of them write it for them selves and out of the goodness of thier heart will put out there for free. They dont need doc's since they wrote it them selves. Be happy that you got it for free. Do you want people to stop releasing code because others complain ? - Original Message - From: John Novack [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, August 15, 2006 12:39 PM Subject: Re: [asterisk-users] Manager Interface API's I, for one, didn't take his comment as anything other than constructive Lack of documentation is an issue, open source or not. It is an unfortunate situation that many very smart coders understand what they have created, but are unwilling or unable to supply enough information for many others to make effective use of their creation How many have struggled through the years with uncommented or poorly commented code when the original creator is off to greener pastures? JMO John Novack Moises Silva wrote: Douglas. Please take this as a constructive comment. I have followed your questions in asterisk-dev and users lists, and you always seem to make non constructive comments about the people giving code/work for Free. And you focus in the negative part, never giving importance to the positive things about it. If you dont like something, then change it yourself, they are not providing a payed service. The source is available AS-IS if you want it, and if you like it, take it; If you dont, just ignore it, try to not make peyorative comments. Regards On 8/15/06, Douglas Garstang [EMAIL PROTECTED] wrote: Well, I don't know about you, but if I have to read the source code to work out how it works, I'm going to go and look at someone elses, that may have some BASIC documentation and examples. -Original Message- From: Don [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 15, 2006 9:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Manager Interface API's Probably cause it is someone like most of us sitting at home doing it...releasing it for free...so why would we write pages of documentation for it? If it's open source and it's free...Then offer them some money to make documentation for it hehe... - Original Message - From: Douglas Garstang To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, August 15, 2006 11:05 AM Subject: [asterisk-users] Manager Interface API's Can anyone recommend the best Manager Interface API, putting language preferences aside? The python and perl ones have bupkiss documentation. I can't understand why anyone would even write an api and make it publically available without documenting it. 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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.10.10/419 - Release Date: 8/15/2006 ___ --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] SIP Qualify
snip The value that qualify takes is the maximum time to accept before considering the device unreachable. If I set qualify to 200ms, and my device's qualify time is 250ms then the device will be considered unreachable. /snip Dosent the phone get this info from asterisk ? Also you are saying that if I have diffrent qualify times set in asterisk and in the phone then it wont 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] SIP Qualify
snip I think you misunderstand what qualify is/does. It appears that you believe that qualify=1000 means that it'll send out a qualify packet every 1000ms. This isn't an unreasonable assumption, but it is wrong. The qualify=1000 means that Asterisk will wait 1000ms for the device to respond to the qualify packet. If after 1000ms there is no yes, I'm here packet, then it will be considered UNREACHABLE. Qualify packets are sent out at a set interval, which, as you can see, is 60 seconds. If the device was previously determined to be UNREACHABLE, the qualify packets will then be sent out every 10 seconds instead. /snip I believed the same. Thats how the docs make it seem. Also is there any way to have them sent out less than every 60 seconds ? ___ --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] IAX unstable with large number of calls?
I was just talking with an unnamed provider and the guy told me that they recommend their users not to use IAX because it is unstable at 50 concurrent calls and unusable at 100 or more calls. Now I have personally worked on an asterisk box that was pushing more than 50 and there were no problems. Anyone else out there have any data either for or against this suggestion? Thanks Curt ___ --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: Page Groups
I have done this with the Citel handsets and it works fine.
--
--
Steven
http://www.glimasoutheast.org
Steve Langstaff [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
For intercom, do you mean placing a call that is automatically answered by the
called party?
If so, the following works for legacy phones connected via a Citel Handset
Gateway, amongst others:
exten = _*803X.,1,Macro(user-callerid)
exten = _*803X.,2,SetVar(_ALERT_INFO=info=alert-autoanswer)
exten = _*803X.,3,SIPAddHeader(Answer-Mode: Auto)
exten = _*803X.,4,Dial(SIP/${EXTEN:4})
(so you dial *803 and then the extension number you want to target)
Similar techniques can be used for page.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Curt Shaffer
Sent: 15 August 2006 17:16
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Page Groups
I have a company that I am going to be moving away from a legacy PBX to
Asterisk. They use page zones pretty heavy and I would like
to keep that functionality. Basically when someone is not at their desk the
receptionist pages all of the phones, telling them there
is a call. Does anyone out there know of the best phones to do this with and if
it is really even possible. I see that intercom is
not supported and paging appears to be minimally supported.
Thanks
Curt
___
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asterisk-users mailing list
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___
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asterisk-users mailing list
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Re: [asterisk-users] Problem with dtmf and voice mail
Never really messed with [EMAIL PROTECTED] Sorry :( - Original Message - From: Paul A Brown [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, August 14, 2006 5:33 PM Subject: Re: [asterisk-users] Problem with dtmf and voice mail Hi Dovid, I can't see how to easily do that in [EMAIL PROTECTED] :-( Any ideas? Paul - Original Message - From: Dovid Bender [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, August 13, 2006 3:44 PM Subject: Re: [asterisk-users] Problem with dtmf and voice mail I had a problem with asterisk real time that if in the general section of sip.conf i was using one form of dtmf and in the real time i set another the dtmf would not work for the first while (dont remember exactly how long). It could be a bug in asterisk. Try making the dtmf in the general section and under that phones setting in sip.conf (or real time) the same and see what happens. Dovid - Original Message - From: Paul A Brown [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, August 11, 2006 6:25 PM Subject: Re: [asterisk-users] Problem with dtmf and voice mail Cheers Dean In extensions config I tried inbound rfc2833 auto info I saved and rebooted phone after each but the problem seemed to stay. Could it be a phone issue? Thanks Paul - Original Message - From: Dean Collins [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, August 11, 2006 5:45 PM Subject: RE: [asterisk-users] Problem with dtmf and voice mail Hi Paul, Happy Friday back. In the config of the extension change the dtmf=XXX Basically there are three ways dtmf can be transmitted by a sip handset, choose another or search the voip-info for the options and you'll solve your problem pretty quickly. Re: sipgatesorry cant help, you'll need to provide more info. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Paul A Brown Sent: Friday, 11 August 2006 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Problem with dtmf and voice mail Hi Guys, Happy Friday I have 2 problems I run [EMAIL PROTECTED] with some Cisco 7960's 1) DTMF - When I dial a number on the 7960 it works fine. However if I dial a number that asks 'Dial 1 for this and 2 for that' and I hit 1 or 2 (or whatever0 the other end acts as though nothing is heard. Any ideas? 2) Voicemail - I use a company called sipgate for my internal route. When someone calls from outsied the call never goes to vmail. However if I dial from ext to ext it does... Any ideas? Thanks Paul ___ --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 ___ --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] 1.2.10 - g726 Issues
Cullin J. Wible wrote: I have hard that 1.2.10 has issues with voice quality through g726. Can anyone provide any feedback or point me in the right direction about the current status of this problem? Been using g726 between multiple * systems for some time and the quality has been very good. Recently, however, all calls via teliax.com using g726 have had very poor quality. Switching back to gsm for them cleared up the iax audio nicely. Not sure if teliax changed something or what, but had been working fine for several months. R. ___ --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] FAX questions
Hi Marco, I'm using T406P(with hardware EC) with a T1-PRI, and I'm having trouble sending fax out though SIP ATA in the same LAN subnet with the Asterisk box. I can send fax out using txfax in call file, but I did have to lower the rxgain and txgain. This is what I'm trying to do: Fax machine --- SIP ATA --LAN-- Asterisk --PRI-- PSTN Have you tried this? Do you have to disable Echo canneler? Thanks. Andy On 8/15/06, Marco Mouta [EMAIL PROTECTED] wrote: Hi, Another question. With latest version of asterisk softwares am I able using rxfax? I had read some remarks about incompatibility between TDM card and rxfax. Is it still exist? I've been using rx for fax reception with TE110P as well as X100P (this only for tests and learning) with very success. As far as i know what could be a problem is that SpanDSP doesn't implements ECM (error correction mode) For Fax reception, only with X100P i've had to play with rxgains, nothing else. I've had some problems only for tx fax lots of errors transmiting faxs, but i think that could be because my * is behind a legacy pbx and i could be facing time sinchronization problems. bye, Zsolt ___ --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] Softphone for Windows Mobile 5?
Hi all, Does anyone know a Softphone for Windows mobile 5? Want to connect to my Asterisk when I am away. Many thanks, Christian ___ --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] FAX questions
Hi,I didn't try that way, only tx fax in call file. But my experience is when u r working with FAX you MUST disable echocanceller!On 8/15/06, Andy Kuo [EMAIL PROTECTED] wrote:Hi Marco, I'm using T406P(with hardware EC) with a T1-PRI, and I'm havingtrouble sending fax out though SIP ATA in the same LAN subnet with theAsterisk box.I can send fax out using txfax in call file, but I did have to lower the rxgain and txgain.This is what I'm trying to do:Fax machine --- SIP ATA--LAN--Asterisk --PRI-- PSTNHave you tried this?Do you have to disable Echo canneler?Thanks. AndyOn 8/15/06, Marco Mouta [EMAIL PROTECTED] wrote: Hi, Another question. With latest version of asterisk softwares am I able using rxfax? I had read some remarks about incompatibility between TDM card and rxfax. Is it still exist? I've been using rx for fax reception withTE110P as well as X100P (this only for tests and learning) with very success. As far as i know what could be a problem is that SpanDSP doesn't implements ECM (error correction mode) For Fax reception, only with X100P i've had to play with rxgains, nothing else. I've had some problems only for tx fax lots of errors transmiting faxs, but i think that could be because my * is behind a legacy pbx and i could be facing time sinchronization problems. bye, Zsolt ___ --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 listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos,Marco Mouta ___ --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: Softphone for Windows Mobile 5?
How about for the blackberry. I know Google talk is supposed to do voice, so maybe I will try that soon. -- -- Steven http://www.glimasoutheast.org Christian [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi all, Does anyone know a Softphone for Windows mobile 5? Want to connect to my Asterisk when I am away. Many thanks, Christian ___ --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: 1 way audio. Dual NIC's.
Are the phones on the same subnet as your server's inside NIC? If not, you will need a manual route added or your server will try sending the audio to the internet, when it should be for an inside phone. Also, verify your DNS. The SIP proxy address on the phone should be pointing to the internal server address. -- -- Steven http://www.glimasoutheast.org William Piper [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] That did the trick. Thanks for the tip. Interesting though. Although technically it is behind a NAT, it is also connecting with the server who is also behind the NAT, I figured that in the eyes of the server... it would need NAT=no because neither device is connecting to it *through* the NAT. Whatever... thanks a million. bp On 8/15/06, Earl Terwilliger [EMAIL PROTECTED] wrote: how about nat=yes qualify=yes canreinvite=no according to: http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions ___ --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] Softphone for Windows Mobile 5?
Try SJphone, it works for me. http://www.sjlabs.com/sjp.html The latency is a little too much over my EVDO cannection though. :) It does work great over wifi. regards, Dave On 8/15/06, Christian [EMAIL PROTECTED] wrote: Hi all, Does anyone know a Softphone for Windows mobile 5? Want to connect to my Asterisk when I am away. Many thanks, Christian ___ --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] New Device
I have spoken to some one who is interested in investing into building equipment for asterisk. I am looking to find out what products that the asterisk community would like to see be built. This can be products that already exists but lack certain functionality as well as things that arent out there but you would want to see it. Thanks. Dovid ___ --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] Cisco 7960 password reset
G'Day List, I am trying, once again, to configure my 7960 to work with asterisk. Where abouts do I go to reset the password on the phone? 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[2]: [asterisk-users] Softphone for Windows Mobile 5?
Hello, Many thanks, but it seems only to be available for Windows Mobile 2003. Will it work on WM5? Many thanks, Christian On 2006-08-15 at 14:00 David Thomas wrote: Try SJphone, it works for me. http://www.sjlabs.com/sjp.html The latency is a little too much over my EVDO cannection though. :) It does work great over wifi. regards, Dave On 8/15/06, Christian [EMAIL PROTECTED] wrote: Hi all, Does anyone know a Softphone for Windows mobile 5? Want to connect to my Asterisk when I am away. Many thanks, Christian ___ --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] 7970 SIP image
Hi Guys, I found a file on the Chisco site for 7970 Sip image (a cop file) but all it had in was xml and png files. No .loads or .sbn Anyone know the exact link? 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] Cisco 7960 password reset
Fastest way (wipes everything out): 1. Power off the phone completely. 2. Hold down the # key, then power the phone on. 3. Continue holding the # key until the LCD gives you a status message. 4. Follow the prompts to do a full factory reset, which resets the password as well. --Maxx Ferguson, Michael wrote: G'Day List, I am trying, once again, to configure my 7960 to work with asterisk. Where abouts do I go to reset the password on the phone? 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 ___ --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 Gizmo?
Hi, How can I use Asterisk and the Gizmo project together? I know that Gizmo is a SIP phone (software - e.g. Not hardware) I want to for example forward a call that I receive from a PSTN line to my Gizmo SIP address, how do I do that? Thank you! Kind Regards, Lennie De Villiers ___ --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] Cisco 7960 password reset
Thanks. Will this action blow away the SIP images I already have on the phone? 'preciate it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Maxx Lobo Sent: Tuesday, August 15, 2006 4:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 password reset Fastest way (wipes everything out): 1. Power off the phone completely. 2. Hold down the # key, then power the phone on. 3. Continue holding the # key until the LCD gives you a status message. 4. Follow the prompts to do a full factory reset, which resets the password as well. --Maxx Ferguson, Michael wrote: G'Day List, I am trying, once again, to configure my 7960 to work with asterisk. Where abouts do I go to reset the password on the phone? 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 ___ --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 Gizmo?
On Tue, Aug 15, 2006 at 10:57:47PM +0200, Lennie De Villiers wrote: Hi, How can I use Asterisk and the Gizmo project together? I know that Gizmo is a SIP phone (software - e.g. Not hardware) I want to for example forward a call that I receive from a PSTN line to my Gizmo SIP address, how do I do that? Their SIP serivce is called sipphone.com . Lookup in their site or in voip-info.org on connecting asterisk to that service. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.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] Cisco 7960 password reset
Maxx, That did not work. Any other ideas? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Maxx Lobo Sent: Tuesday, August 15, 2006 4:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 password reset Fastest way (wipes everything out): 1. Power off the phone completely. 2. Hold down the # key, then power the phone on. 3. Continue holding the # key until the LCD gives you a status message. 4. Follow the prompts to do a full factory reset, which resets the password as well. --Maxx Ferguson, Michael wrote: G'Day List, I am trying, once again, to configure my 7960 to work with asterisk. Where abouts do I go to reset the password on the phone? 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 ___ --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] Cisco 7960 password reset
What Cisco image is the phone running? If it is really old (lower than P0S030203) then yeah, this won't work. If you upgrade the phone to P0S30203 and from there on to P0S3-06-3-00, and then these instructions will work fine. This should be pretty straightforward using ATFTP and the Cisco images. In response to your other question, a factory reset TMK does not wipe out the SIP image. Just the settings. --Maxx Ferguson, Michael wrote: Maxx, That did not work. Any other ideas? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Maxx Lobo Sent: Tuesday, August 15, 2006 4:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 password reset Fastest way (wipes everything out): 1. Power off the phone completely. 2. Hold down the # key, then power the phone on. 3. Continue holding the # key until the LCD gives you a status message. 4. Follow the prompts to do a full factory reset, which resets the password as well. --Maxx Ferguson, Michael wrote: G'Day List, I am trying, once again, to configure my 7960 to work with asterisk. Where abouts do I go to reset the password on the phone? 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 ___ --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] STRFTIME dialplan function not picking up system timezone
Hi all,
I've just been playing with the STRFTIME dialplan function and am having
trouble getting it to pickup my systems local timezone.
According to show function STRFTIME and voip-info.org all the arguments are
optional and according to voip-info.org if you leave them out they will
default to the current time, the current timezone and %c respectively.
My local timezone is Pacific/Auckland (GMT+12) which is setup correctly
AFAIK - date returns the correct time and timezone. I have also tried setting
TZ=Pacific/Auckland and running asterisk at that console which didn't alter
the behaviour.
If I call a test extension with this in the dialplan;
NoOp(${STRFTIME(,,)}))
NoOp(${STRFTIME(,Pacific/Auckland,)}))
then I get this output (shortened) ;
NoOp(SIP/800-081778a4, Tue Aug 15 22:11:36 2006))
NoOp(SIP/800-081778a4, Wed Aug 16 10:11:36 2006))
I have also tried reading asterisk/stdtime/localtime.c which is (I think)
where this stuff goes on but it's over my head.
Does anyone have any ideas as to why I can't get this to work or am I
expecting the wrong behaviour (using SVN trunk)?
Cheers,
hads
--
http://nicegear.co.nz
New Zealand's VoIP supplier
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[asterisk-users] Multiple registrations to the same asterisk server
Hi All, I have the following scenario: A central Asterisk server where
all the ATAs register themselves. This server runs Asterisk 1.2.5 and
ATAs are SPA-2002. So far everything is OK. Now I have another location
where I want to connect 4 analog phones. I thought setting up 2 SPA-2002
but as I already have a TDM400P card and I want to use it, I had
configured asterisk 1.0.7 on the second machine. So far I can place
calls from the second server to any extension on the central server. But
I cant get an ATA on the central server to reach an extension on the
second server. Please help me solve this situation. Thanks in advance.
Juan Luis Moyano
The configs are as follows:
Central Server
--
-sip.conf
[40019]
username=USER1
callerid=40019
type=friend
host=dynamic
secret=
mailbox=40019
accountcode=USER1
[40028]
username=USER2
callerid=40028
type=friend
host=dynamic
secret=
mailbox=40028
accountcode=USER2
[4]
username=USER3
callerid=4
type=friend
host=dynamic
secret=
mailbox=4
accountcode=USER3
[40023]
username=USER4
callerid=40023
type=friend
host=dynamic
secret=
mailbox=40023
accountcode=USER4
Second Server
-
-sip.conf
register = 40019:[EMAIL PROTECTED]/40019
register = 40028:[EMAIL PROTECTED]/40028
register = 4:[EMAIL PROTECTED]/4
register = 40023:[EMAIL PROTECTED]/40023
[40019]
type=friend
secret=
username=40019
host=10.32.1.16
insecure=very
[4]
type=friend
secret=
username=4
host=10.32.1.16
insecure=very
[40028]
type=friend
secret=
username=40028
host=10.32.1.16
insecure=very
[40023]
type=friend
secret=
username=40023
host=10.32.1.16
insecure=very
-extensions.conf
[globals]
USER1=Zap/2
USER2=Zap/3
USER3=Zap/4
USER4=Zap/5
[extensions]
exten = 40019,1,Dial(${USER1})
exten = 40023,1,Dial(${USER2})
exten = 40028,1,Dial(${USER3})
exten = 4,1,Dial(${USER4})
[outbound]
exten = _.,1,Dial(SIP/[EMAIL PROTECTED])
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Re: Re[2]: [asterisk-users] Softphone for Windows Mobile 5?
Yes, use it on WM5. Dave On 8/15/06, Christian [EMAIL PROTECTED] wrote: Hello, Many thanks, but it seems only to be available for Windows Mobile 2003. Will it work on WM5? Many thanks, Christian On 2006-08-15 at 14:00 David Thomas wrote: Try SJphone, it works for me. http://www.sjlabs.com/sjp.html The latency is a little too much over my EVDO cannection though. :) It does work great over wifi. regards, Dave On 8/15/06, Christian [EMAIL PROTECTED] wrote: Hi all, Does anyone know a Softphone for Windows mobile 5? Want to connect to my Asterisk when I am away. Many thanks, Christian ___ --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: Re[2]: [asterisk-users] Softphone for Windows Mobile 5?
Sorry, poor reply. Yes I use it on WM5, and have not seen any problems. I admit I don't use it a lot, but it does seem to work fine. regards, Dave On 8/15/06, David Thomas [EMAIL PROTECTED] wrote: Yes, use it on WM5. Dave On 8/15/06, Christian [EMAIL PROTECTED] wrote: Hello, Many thanks, but it seems only to be available for Windows Mobile 2003. Will it work on WM5? Many thanks, Christian On 2006-08-15 at 14:00 David Thomas wrote: Try SJphone, it works for me. http://www.sjlabs.com/sjp.html The latency is a little too much over my EVDO cannection though. :) It does work great over wifi. regards, Dave On 8/15/06, Christian [EMAIL PROTECTED] wrote: Hi all, Does anyone know a Softphone for Windows mobile 5? Want to connect to my Asterisk when I am away. Many thanks, Christian ___ --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] Multiple registrations to the same asterisk server
Hi , Please post here your extensions.conf in your central server only with that i can figured out or at least try to help u.Best regards,Marco MoutaOn 8/15/06,
Juan Luis Moyano [EMAIL PROTECTED] wrote:
Hi All, I have the following scenario: A central Asterisk server whereall the ATAs register themselves. This server runs Asterisk 1.2.5 andATAs are SPA-2002. So far everything is OK. Now I have another location
where I want to connect 4 analog phones. I thought setting up 2 SPA-2002but as I already have a TDM400P card and I want to use it, I hadconfigured asterisk 1.0.7 on the second machine. So far I can placecalls from the second server to any extension on the central server. But
I cant get an ATA on the central server to reach an extension on thesecond server. Please help me solve this situation. Thanks in advance.Juan Luis MoyanoThe configs are as follows:Central Server
---sip.conf[40019]username=USER1callerid=40019type=friendhost=dynamicsecret=mailbox=40019accountcode=USER1[40028]username=USER2callerid=40028
type=friendhost=dynamicsecret=mailbox=40028accountcode=USER2[4]username=USER3callerid=4type=friendhost=dynamicsecret=mailbox=4accountcode=USER3
[40023]username=USER4callerid=40023type=friendhost=dynamicsecret=mailbox=40023accountcode=USER4Second Server--sip.confregister =
40019:[EMAIL PROTECTED]/40019register = 40028:[EMAIL PROTECTED]/40028register = 4:[EMAIL PROTECTED]/4register = 40023:[EMAIL PROTECTED]/40023[40019]type=friendsecret=
username=40019host=10.32.1.16insecure=very[4]type=friendsecret=username=4host=10.32.1.16insecure=very
[40028]type=friendsecret=username=40028host=10.32.1.16insecure=very[40023]type=friendsecret=username=40023host=
10.32.1.16insecure=very-extensions.conf[globals]USER1=Zap/2USER2=Zap/3USER3=Zap/4USER4=Zap/5[extensions]exten = 40019,1,Dial(${USER1})exten = 40023,1,Dial(${USER2})
exten = 40028,1,Dial(${USER3})exten = 4,1,Dial(${USER4})[outbound]exten = _.,1,Dial(SIP/[EMAIL PROTECTED])___--Bandwidth and Colocation provided by
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-- Com os melhores cumprimentos,Marco Mouta
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[asterisk-users] macro-dialout without specifying trunk
I am trying to have a SIP extension that will dial an outside phone
number (ie: cell phone) using a zap channel.
I am using the following hack, which doesn't technically works, but not
nicely. What i want to do is have it pick an available trunk from zap1
to zap20.
I have tried using dialout,s,number and also dialout,g1,number i
just keep getting all circuits busy. (I have posted my zapata.conf below).
I can do it if I specify the specific trunk. Here is my extension:
exten = 299,1,Macro(dialout-return,1,1914426)
exten = 299,2,Macro(dialout-return,2,1914426)
exten = 299,3,Macro(dialout-return,5,1914304)
exten = 299,4,hangup
; dialout-return. Like dialout but doesn't go to outisbusy.
[macro-dialout-return]
exten = s,1,GotoIf($[foo${ECID${CALLERIDNUM}} = foo]?4);check
for CID override for exten
exten = s,2,SetCallerID(${ECID${CALLERIDNUM}})
exten = s,3,Goto(6)
exten = s,4,GotoIf($[foo${OUTCID_${ARG1}} = foo]?6);check
for CID override for trunk
exten = s,5,SetCallerID(${OUTCID_${ARG1}})
exten = s,6,SetVar(length=${LEN(${DIAL_OUT_${ARG1}})})
exten = s,7,Dial(${OUT_${ARG1}}/${ARG2:${length}})
zapata.conf
[trunkgroups]
[channels]
language=en
context=from-pstn
signalling=pri_cpe
switchtype=national
rxwink=300 ; Atlas seems to use long (250ms) winks
callerid=asreceived
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=400
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
faxdetect=incoming
channel = 1-7
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Re: [asterisk-users] PRI Dropouts (Solved)
Kevin Withnall wrote: After much playing and getting nowhere, I was on the phone to the guys from www.voipshop.com.au and mentioned that the pri dropout problem was occuring and if they had any solutions. Immediately they mentioned something that causes a problem in australia. On longdistance phone calls (sometimes) you hear a series of short beeps that indicates to the receiver that it's a long distance call. Few calls seem to do this these days but some do. It seems to be exactly what the busy detect code looks for. On removing the busydetect (or rather setting it to 0) it solved our dropout problems. This is rather strange. AFAIK, busydetect is only applicable to analog FXO ports. PRI doesn't depend on call progress tones, everything is in the D-channel. Can anyone else confirm that busydetect is used for PRI in chan_zap? If so, I would consider it a bug. Cheers. Leo ___ --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 reject a call without picking it up, (E1-T1-ISDN)
Hi, I´m in a bit of a hurry here, I need to reject calls before picking them up. If I do hangup on the first line, does anyone knows if the line counts as picked up for the Telco? how about if I register the incoming callerid, and then do hangup on the second line? 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
[asterisk-users] New asterisk jukebox needs testing
Asterisk people, I just wrote a new version of my Asterisk Jukebox application in C. Testers and feedback is very much welcome and appreciated! http://www.lobstertech.com/code/jukebox/ Here is the description on the site: Asterisk Jukebox is an IVR application written for Asterisk, an open source PBX application. Asterisk Jukebox allows a caller to browse your music collection. All you have to do is tell Jukebox where your music is and callers will be able to browse the collection. For example a caller might hear: Press 1 for Sisters Of Mercy, (Caller presses 1), Press 1 for Marian, Press 2 for This Corrosion, etc. But that's not all! You can also tell the Jukebox to automatically pick random songs to play. At the moment, the jukebox supports all the standard Asterisk formats such as GSM and uLaw. There is also MP3 support if you have CPU to spare and mpg123 installed via app_mp3. You must also have Festival installed on your system so the Jukebox can generate text to speech. - Justin Tunney ___ --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] g.711 Codec Question
Dave, QAs far as I remember it, it goes like this... if your inside phone is *only* G711-alaw and your trunk (SIP or IAX) is *only G711-ulaw that at session negotiation (call setup) Asterisk will woprk out that it has to remain in the loop and transcode. If you run-up asterisk and bother to watch the start-up debug messages then it shows you the relative costs (in terms of CPU utilisation per call) for the various transcoding between codecs. From recollection G.711-Alaw to G.711-Ulaw is relatively inexpensive with a cost of 1 ... Regards Mike - Original Message - From: David Thomas [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, August 14, 2006 8:57 PM Subject: [asterisk-users] g.711 Codec Question Greeting Everyone, I don't have access to Asterisk box right now or I'd check this myself... If my client phone uses g.711 (alaw) and my outbound trunk leaving asterisk uses g.711 (ulaw), will asterisk have to transcode? If so is the processing overhead much? regards, Dave ___ --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] AsteriskSpeaksGoogleTalk - User is always disconnected - Problems
Hi,I've just followed http://www.voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talk instructions from wiki,And i always get my jabber (GoogleTalk account for asterisk server) not registred: asterisk1*CLI jabber show connectedJabber Users and their status: User: [EMAIL PROTECTED] - Disconnectedasterisk1*CLI jabber testUser: [EMAIL PROTECTED]Oooh a working message stack!-- jabber.conf[general]debug=yesautoprune=yesautoregister=yes[asterisk]type=clientserverhost=talk.google.comusername= [EMAIL PROTECTED]secret=port=5222usetls=yesusesasl=yesbuddy=[EMAIL PROTECTED]statusmessage=I am an Asterisk Server timeout=100-jingle.conf[general]context=defaultallowguest=yes[guest]disallow=allallow=ulawcontext=guest[marco.mouta ]username=[EMAIL PROTECTED]disallow=allallow=ulawcontext=googletalkconnection=asterisksip.conf [general]context=googletalkbindport=5060bindaddr=0.0.0.0srvlookup=yesdtmfmode=rfc2833relaxdtmf=nodisallow=allallow=ulawallow=alawallow=gsmmaxexpirey=30 defaultexpirey=180canreinvite=yesnat=0UserAgent=Asteriskechocancel=yesechocancelwhenbridge=yesCan any one help me on this? It may also help if you can explain me the relation between those files jingle.conf and jabber.conf, i mean who is who, and their goals.-- Com os melhores cumprimentos,Marco Mouta ___ --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] Cisco 7960 password reset
Maxx, Thanks much for the feedback. I will check into it and follow up with your instructions. 'preciate it. Best wishes. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Maxx Lobo Sent: Tuesday, August 15, 2006 5:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 password reset What Cisco image is the phone running? If it is really old (lower than P0S030203) then yeah, this won't work. If you upgrade the phone to P0S30203 and from there on to P0S3-06-3-00, and then these instructions will work fine. This should be pretty straightforward using ATFTP and the Cisco images. In response to your other question, a factory reset TMK does not wipe out the SIP image. Just the settings. --Maxx Ferguson, Michael wrote: Maxx, That did not work. Any other ideas? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Maxx Lobo Sent: Tuesday, August 15, 2006 4:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 password reset Fastest way (wipes everything out): 1. Power off the phone completely. 2. Hold down the # key, then power the phone on. 3. Continue holding the # key until the LCD gives you a status message. 4. Follow the prompts to do a full factory reset, which resets the password as well. --Maxx Ferguson, Michael wrote: G'Day List, I am trying, once again, to configure my 7960 to work with asterisk. Where abouts do I go to reset the password on the phone? 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 ___ --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] Abstraction for a newbie
Generally, the turn off voicemail function you used tells the provider *what to do when they can't get ahold of your asterisk* -- this doesn't typically mean 'send all calls to my voicemail until I turn this feature off' All calls should attempt to contact your asterisk server (or at least check if it's registered recently) before sending the caller to voicemail. As Mike pointed out in another post, this issue is almost certainly either with your sip.conf or your firewall config. Moj Dominic Son wrote: Hi. Can someone explain to a right brained person what is going on with In/out bound trunks, how it connects to my Trixbox.. 1. i get issued a free NY phone number from a voip service like stanaphone . 2. i then call this number, it connects to the stanaphone voicemail 3. i turn off the voicemail because i want it to connect to my Askterisk, I've set up all the trunks in the PBX setup, ( sip.stanaphone, etc) 4. now i call my NY number, and it says 'this phone is not in service, please check the number and dial again' my Q: how does this work, more specifically, if i turned off the VM, how does stanaphone then know to look for my asterisk server to use the trixbox? -- Anything else, let me know. !DSPAM:500,44dd080a25292693510148! ___ --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 !DSPAM:500,44dd080a25292693510148! -- Mojo [EMAIL PROTECTED] Office Manager, Horan Company, LLC (907) 747- x112 ___ --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 reject a call without picking it up, (E1-T1-ISDN) In a Hurry
Basically, if I do hangup on the first line the console shows: Starting simple switch on Zap/2-1 Accepting overlap call from '' to '3423' on channel 0/1, span 1 executing Hangup (Zap/2-1, ) in new stack. I believe this is actually picking the call up isn't it? Manrique Feoli escribió: Hi, I´m in a bit of a hurry here, I need to reject calls before picking them up. If I do hangup on the first line, does anyone knows if the line counts as picked up for the Telco? how about if I register the incoming callerid, and then do hangup on the second line? 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 ___ --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 host and registering
I have a setup where in sip.conf the host=ser.zxy.com for the phones. Non of the phones are connected to Asterisk directly, but are connected to SER. Thus non of the phones registry with Asterisk. I have noticed that when I forward a call to Asterisk it doesn't send the call back to SER (which is very good in my scenario). I was wondering if this was intended behaviour for Asterisk, that a call only gets sent to a phone if its registered (e.g. this behaviour wont change come the next version of Asterisk) ? The reason I have set host to SER is so that the MWI is sent to SER, but I don't want a call sent from SER to Asterisk to be sent back to SER. Thanks -- 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
[asterisk-users] Just a test
It's justa test sorry. Edgar Alonso Lopez Chavez ESIME IPN ___ --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] anothet tes
otra prueba Edgar Alonso Lopez Chavez ESIME IPN ___ --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] modprobe wctdm fails in /etc/rc.local on FC5
If I boot my server and manually type modprobe wctdm, it correctly loads both wctdm and zaptel. If I put the modprobe in /etc/rc.local and reboot, it fails. Why? I am running the latest svn source of zaptel on Fedora Core 5 (w/latest updates as of 8/15) Here are the error messages from modprobe: Aug 15 15:55:39 WARNING[1860] chan_zap.c: Unable to specify channel 1: No such device or address Aug 15 15:55:39 ERROR[1860] chan_zap.c: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Aug 15 15:55:39 ERROR[1860] chan_zap.c: Unable to register channel '1' Aug 15 15:55:39 WARNING[1860] loader.c: chan_zap.so: load_module failed, returning -1 Aug 15 15:55:39 WARNING[1860] loader.c: Loading module chan_zap.so failed! ___ --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: modprobe wctdm fails in /etc/rc.local on FC5
Can someone send me their modprobe.conf file? I think that may be the problem. A zaptel file is created during install in /etc/ modprobe.d but modprobe.conf must need to reference 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
[asterisk-users] Question about queue
Hi, I have set a queue 5000 with a agent logged in. When an user dial 5000, will ring and then answer the call. Queue function will execute and recording is started. However, the recorded wav file is not a valid file with just a few bytes. Anyone can help me to enable the recording function in queue? -- Executing MacroExit(SIP/2001-006bd8e0, ) in new stack -- Executing Goto(SIP/2001-006bd8e0, sipcom|5000|1) -- Goto (sipcom,5000,1) -- Executing Answer(SIP/2001-006bd8e0, ) -- SIP Seeding peer from astdb: '2001' at [EMAIL PROTECTED]:5060 for 120 -- Executing Set(SIP/2001-006bd8e0, MONITOR_FILENAME=1155691672.29) -- Executing Queue(SIP/2001-006bd8e0, 5000|t) -- Started music on hold, class 'default', on channel 'SIP/2001-006bd8e0' -- Called Local/[EMAIL PROTECTED] -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/|40) In CDR, there are two records about the queue. lastapp,lastdata,duration,billsec,disposition,uniqueid 1) Queue, 5000|t, 232,232,ANSWERED,1155691672.29 2) Dial, SIP/|40,26,0,ANSWERED,1155691672.31 In the monitor folder, there is a file 1155691672.29.WAV with size 190. ___ --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] Polycom upgrade issue
OK, I may have done something stupid. I was trying to upgrade my Polycom to the newest firmware I could find (1.6.7). I am also trying to get provisioning working from a central server. I tired to reset with holding 468* down and it kept the settings the phone had on the phone. From what I understand the settings on the phone override all. So I went into reset it from the phone and choose to format the firmware. Now when I try to boot it I am getting the following in the *-boot.log 0527180621|cfg |4|00|Could not get all 512 bytes of the header. 0527181013|cfg |4|00|Could not get all 512 bytes of the header. 0527181014|app1 |6|00|Error application is not present. 0527181014|app1 |6|00|Uploading boot log, time is SAT MAY 27 18:10:14 2006 I tried to put the old firmware and configs back in the directory but I get the same thing. Any help out there? Thanks! Curt ___ --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
