Re: [asterisk-users] [SOLVED] PlayDTMF and Asterisk Manager
Thanks for your response :-). (I'm in GMT+2, and I currently have no internet at home, so that is why my response is so late) On second thought--- it would be silly to make this sort of application! When would you run it? The feature/wish/need is for when using both a CTI software and a IP phone, and avoid unnecessary handlings for agents : all the agents will use only the CTI (and not a softphone :-() in order to act upon the IP phone. So I could need this feature when an agent will want to make an attended transfer of an call to someone else (I didn't figure out how to do otherwise [1]). Could I do it otherwise ? [1] maybe with the bridge function from 1.6, I will be able to do so _ Ten : Messenger en illimité sur votre mobile ! http://mobile.live.fr/messenger/ten/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Transfer Question
I'm having a tough time figuring out how to do something. If I have an operator (which could potentially be in their own context) and an internal-only context, is it possible to make it so the operator can call the internal-only context but *NOT* transfer calls to it? The idea is that the internal-only context should not be allowed to make or receive outside calls. The only concern is that the operator and other office users can transfer outside calls to these internal-only extensions. Also, the operator and office extensions need to be able to call the internal-only extensions directly. Thanks! Mark ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [SOLVED] PlayDTMF and Asterisk Manager
It can be done - I saw a tech do it the other day. They used the 'local' dial option, with the D option (from memory) If you want more info, I can grab it from them next week PaulH On Thu, 2007-07-12 at 12:54 +, lemmel lemmel wrote: So I'm back on this matter (I thus give enough time for a good samartan to help), and now, I think that PlayDTMF is not designed for what I want to do : I wanted to simulate pressed keys in order to give order to Asterisk (an attented xfert) ; e.g. generate a #2 (it is my combination to performe an attended transfer) via Asterisk on the agent channel, and hope that Asterisk will deal this generation as order to himself (like those keys was indeed pressed by the agent). But I think that those DTMF simulation are intented to the guy at the end of the channel, and not asterisk. _ Gagnez des pc Windows Vista avec Live.com http://www.image-addict.fr/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue property
Yes, you are right. I know that I´m trying to use a queue with a strange functionality. Life is hard!! I try to combine a short timeout and a short maxlimit. Thanks a lot!! On 7/12/07, Anthony Francis [EMAIL PROTECTED] wrote: Carlos Chavez wrote: On Thu, 2007-07-12 at 15:03 -0300, equis software wrote: I try setting joinempty=strict, but if there are all agent busy answering a call, i need that the queue reject new callers until one of the agents ends their calls. Example: Agent1 on call Agent2 on call When one call arrive, the queue reject the call. I don´t want to have callers waiting in the queue. Thanks! Than why have a queue at all? The function of a queue is to have more callers than agents. You can set the parameter maxlimit=2 on the queue so if you only have two agents it will not allow any more caller to join the queue. Again, this defeats the purpose of a queue. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I believe maxlimit defines calls waiting. I could be wrong, I didn't take time to confirm that. I agree, thought you should just use dial(sip/dev1sip/dev2,20,tr) or similiar and then if the agents are not available you can do whatever else with the call. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QUEUE_WAITING_COUNT
[EMAIL PROTECTED] wrote: Hi, I'm playing around with the QUEUE_WAITING_COUNT function but it always seems to return zero? I've tried everything. I suspect that this feature is not implemented in 1.2.7 which I am running.. Does anyone know in which version this function was added? Regards, Jan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users That function is not available in 1.2. It is in all versions of 1.4 and trunk. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] no ringback from SIP server when originating call
I have an application that uses the Asterisk Management Interface to bridge two calls using the Originate command with Dial as the action. Using one SIP server, there is no ringback on the second leg of the call. The first person is called, answers, and hears silence until the second person picks up, even though the second person's phone is ringing. When the call goes to another SIP gateway, ringback works fine. From SIP traces I found that the one that works returns 180 ringing to Asterisk and the one that doesn't work returns 100 trying followed by 183 session progress. It is my understanding that 180 ringing causes ringback to be generated by the callee, while 183 means that the caller has early media and will send ringback through RTP. Anyone have any idea why I wouldn't get ringback in this case? Should Asterisk be passing through the early media to the first caller even though the second caller has not answered? I am not using the r option to the Dial command. I have tried it both on and off and get no ringback in either case. I have also tried variations of the progressinband setting. I have listened to the RTP going from the SIP server to Asterisk and I can hear the ringing in it. It seems like Asterisk isn't sending any audio to the first caller until both parties answer. Thanks, Matthew Boedicker ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] limit simultaneous calls
Hi, is there a way to limit an account to do simultaneous calls in sip and iax? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [SOLVED] PlayDTMF and Asterisk Manager
They used the 'local' dial option, with the D option (from memory) -- Documentation : D(digits): After the called party answers, send digits as a DTMF stream, then connect the call to the originating channel. (You can also use 'w' to produce .5 second pauses.) -- The Dial D option is for a dialplan command and is not conditionnal (the agent may or may not want a transfert) and is designed in order to emit DTMF on the called channel (e.g. in order to pass through a IVR) before the channels are bridged, and not in order to give to asterisk order. If you want more info, I can grab it from them next week Why not, but I don't think that will help me. _ Avec Windows Live OneCare éliminez tous les virus de votre PC ! http://www.windowslive.fr/liveonecare/default.asp ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] limit simultaneous calls
On Fri, 2007-07-13 at 21:58 +0800, Mark Quitoriano wrote: is there a way to limit an account to do simultaneous calls in sip and iax? You can use the GROUP and GROUP_COUNT dialplan functions to enforce arbitrary limits as you see fit. There's an example on the wiki at http://www.voip-info.org/wiki/view/Asterisk+Queue+with+limited+calls+per +IAX+agent -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk-addons compilation error: dereferencing pointer to incomplete type
I am having trouble getting asterisk-addons 1.4.2 to compile (after a
successful configure). Asterisk itself (and AsteriskGUI) compile fine.
I get:
cdr_addon_mysql.c: In function `handle_cdr_mysql_status':
cdr_addon_mysql.c:91: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:93: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:95: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:97: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:98: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:99: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:100: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c: In function `mysql_log':
cdr_addon_mysql.c:154: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:155: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:155: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:155: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:155: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:220: error: dereferencing pointer to incomplete type
...lots more of these
cdr_addon_mysql.c:222: error: dereferencing pointer to incomplete type
...lots more of these
cdr_addon_mysql.c:225: error: dereferencing pointer to incomplete type
...lots more of these
cdr_addon_mysql.c:227: error: dereferencing pointer to incomplete type
...lots more of these
cdr_addon_mysql.c: In function `my_load_config_string':
cdr_addon_mysql.c:279: warning: assignment makes pointer from integer
without a cast
cdr_addon_mysql.c: In function `my_load_module':
cdr_addon_mysql.c:378: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:382: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:383: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:384: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:385: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:387: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:388: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:389: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:390: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:391: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:392: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:393: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:394: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:395: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:396: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:397: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:398: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:399: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:400: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:401: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:408: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:409: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:409: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:409: error: dereferencing pointer to incomplete type
cdr_addon_mysql.c:409: error: dereferencing pointer to incomplete type
make[1]: *** [cdr_addon_mysql.o] Error 1
make: *** [all] Error 2
Has anyone come across this?
TIA
--
Jeremy Malcolm LLB (Hons) B Com
Internet and Open Source lawyer, IT consultant, actor
host -t NAPTR 1.0.8.0.3.1.2.9.8.1.6.e164.org|awk -F! '{print $3}'
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Re: [asterisk-users] Channel name in queue log replaced by a manager event?
On 7/12/07, Jared Smith [EMAIL PROTECTED] wrote: It probably wouldn't hurt to open a bug for this... I've seen something like this before, only it was manager events ending up inside of SIP traffic. It definitely sounds like a pointer problem or maybe a locking problem to me... which means it's probably going to be difficult to track down. Filed as 10199, with a bit more info about the queue config and dialplan being used to enqueue callers. Thanks -- j. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to load phone registration information
I'm using realtime sip already! To let you understant better my problem, I'll explain a bit more: In a redundancy scheme, I have two asterisk servers, each running on different machines although sharing the same MySQL DB for relatime sip. Problem arises when the second server assumes the production. When some phone tries to establish a new call, those INVITEs reach the new server, although this server seems to don't read the registration information kept in sip_buddies table to know if the destination phone is registered or not, and so, the call fails. Because the destination phone was registered in the first server, I was expecting that the second server when assuming production would first read the sip_buddies DB table to see if the destination phone was registered or not, but that seems to don't happen. It seems that registration information is only kept in memory and isn't read from DB! Is there any way that I can force Asterisk to read sip_buddies realtime DB table to know if destination phone is registered? Regards, Ricardo Carvalho. On 7/12/07, Ricardo Carvalho [EMAIL PROTECTED] wrote: Is it possible to load phone registration information stored in sipfriends MySQL DB, so that Asterisk thinks those phones are already registered? This would be very usefull for a redundant server... Look at realtime sip should help you ram ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 on Solaris SPARC/x86/x64 Codec
There already are x86 Solaris builds for codec_g729 - ftp.digium.com/pub/telephony/codec_g729/unsupported/ - Bruce McAlister [EMAIL PROTECTED] wrote: Hi All, Does anyone know what the current status is of the G729 codec on Solaris? According to the following link: http://www.asteriskvoipnews.com/asterisk_releases/_digium_g729_codec_now_available_forsolarissparc.html there is a version available for SPARC processor's. However, I have just had a quick look around Digium's FTP server and cannot seem to find these codecs (supported or unsupported). Does anyone know if Digium plan on releasing a SPARC *and/or* Intel/AMD G729 codec on Solaris? I would have thought with the availability of Solaris and Open Solaris that a little more enthusiasm would have been forthcomming in getting the codecs running on those environments? Thanks Bruce ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jason Parker Digium ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] QUEUE_WAITING_COUNT
Hi, I'm playing around with the QUEUE_WAITING_COUNT function but it always seems to return zero? I've tried everything. I suspect that this feature is not implemented in 1.2.7 which I am running.. Does anyone know in which version this function was added? Regards, Jan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk snmp
Hello, I'm trying to monitor asterisk with snmp. I'm using asterisk 1.4.4 compiled with res_snmp on a debian stable: *CLI module show like snmp Module Description Use Count res_snmp.soSNMP [Sub]Agent for Asterisk 0 I've configured asterisk in res_snmp.conf: [general] subagent = yes enabled = yes and when asterisk start print this: [Jul 13 09:14:58] VERBOSE[3330] logger.c: == Parsing '/etc/asterisk/res_snmp.conf': [Jul 13 09:14:58] VERBOSE[3330] logger.c: Found [Jul 13 09:14:58] VERBOSE[3330] logger.c: res_snmp.so = (SNMP [Sub]Agent for Asterisk) at the snmp I have added the agentx support and asterisk in snmpd.conf: master agentx agentXperms 0660 0550 root root access asteriskany noauthexact allnone none group asterisk v1 paranoid I've added the option -x /var/agentx/master at snmp start. pizov:/etc/snmp# ps aux | grep snmp snmp 4004 0.0 0.9 8420 4692 ?S10:51 0:00 /usr/sbin/snmpd -Lsd -Lf /dev/null -u snmp -I -smux -p /var/run/snmpd.pid 127.0.0.1 -x /var/agentx/master -V root 4007 0.0 0.3 7724 1828 ?Ss 10:51 0:00 /usr/sbin/snmptrapd -a -Lsd -p /var/run/snmptrapd.pid I've copied the mibs files: cp doc/asterisk-mib.txt /usr/share/snmp/mibs/ cp doc/digium-mib.txt /usr/share/snmp/mibs/ and restart the snmpd. In the logs of snmp: Jul 13 11:53:49 localhost snmpd[4132]: cache has existing timer id. Jul 13 11:53:50 localhost snmpd[4132]: Turning on AgentX master support. Jul 13 11:53:50 localhost snmpd[4132]: NET-SNMP version 5.2.3 and when I start snmp in the asterisk console appears: *CLI NET-SNMP version 5.2.3 AgentX subagent connected but when I try to walk throught the asterisk snmp: pizov:/usr/share/snmp/mibs# snmpwalk -v 1 -c public localhost ASTERISK-MIB::astVersionTag End of MIB pizov:/usr/share/snmp/mibs# snmpwalk -v 1 -c public localhost asterisk asterisk: Unknown Object Identifier (Sub-id not found: (top) - asterisk) What am I doing wrong?? Thanks a lot Roger -- Roger Casaposna - Adam Telefonía IP email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] www: http://www.adamvozip.es http://www.adamvozip.es/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Distribution lists for voicemail
Hi, I am sure someone has already asked this, but I am fairly new to the list, so I haven't seen anything on this. Does anyone know of a good way to leave one voicemail message, and the message be forwarded to multiple voicemail boxes at once. I realize that we could leave one message in a group box and have the message forwarded to a group email address, but their are some users that want the message in their actual voicemail box. Thank you for your help. Jadrien Wauthier ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Micros-Fidelio - billing in hotel
Lee Jenkins wrote: I'd say that Micro is the MS of Restaurant POS. We replace their systems regularly ;) I'm curious what with? -- Chris Mason Anguilla: (264) 497-5670 Fax: (264) 497-8463 Cell: 264-235-5670 International: (305) 704-7249 Fax: (815)301-9759 Yahoo IM only: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH stop and resume when i hold
2007/7/12, Noah Miller [EMAIL PROTECTED]: Hi - I have a strange comportment of the MOH system on my asterisk. When i respond to a call and after fews second i set this call in hold mode the correspondent listen the music fine. When i re-take my correspondent at T0 instant the music is paused. And when i re-hold him at T60 (60 second later) the sound is always at T0 when he was stopped at T0. So the music is stopped and don't continue (normal comportment) when i speak with peoples. Someone know the problem? a solution? What you're describing is the asterisk native files method for MOH. This behavior is the way it was designed. If you want continuous MOH, I'd suggest using a third-party player like madplay or mpg123. You can read more here: http://www.voip-info.org/wiki/index.php?page=Asterisk%20config%20musiconhold.conf (Hint: Scroll down to the section labeled Pre-Asterisk 1.2 Information Below) - Noah Hi Noah, Thanks for your answer. I have tested with all the method : mpg123, madplay, mp3 native in asterisk (asterisk-addons). And the result is the same for all this tests: the music is stopped and resumed after. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems with RNDIS
Hello List I am having some problems receiving RNDIS on a EuroISDN E1 in both Asterisk 1.2 and 1.4. Im not receiving anything, and when I do a pri debug span, I get this message: -- Making new call for cr 114 -- Processing Q.931 Call Setup -- Processing IE 161 (cs0, Sending Complete) -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 28 (cs0, Facility) Handle Q.932 ROSE Invoke component [ Handling operation 15 ] !! Unable to handle ROSE operation 15 [ 30 19 02 01 01 0A 01 02 A1 11 A0 0F A1 0D 0A 01 02 12 08 32 32 34 35 38 34 30 35 ] - [0..22458405] -- Processing IE 108 (cs0, Calling Party Number) -- Processing IE 112 (cs0, Called Party Number) q931.c:3294 q931_receive: call 114 on channel 28 enters state 6 (Call Present) q931.c:2570 q931_call_proceeding: call 114 on channel 28 enters state 9 (Incoming Call Proceeding) The 22458405 is the RDNIS that is supposed to be in the RDNIS field. Can anybody see why this is? Is it our operator that sends the information incorrectly? Kind Regards Jon Leren Schøpzinsky No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.4/898 - Release Date: 12-07-2007 16:08 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lines Not being Hung UP Major
Can't help you with the cause but I can tell you that you can use the soft hangup command to kill those channels instead of restarting. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael J. Liberatore Sent: Thursday, July 12, 2007 3:56 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Lines Not being Hung UP Major Hi all, i am having a major asterisk problem. I think it started around 1.2.19 but could also be 1.2.18 zaptel or 6.5.10 snom 360. basically we start getting busy signals, all our 4 line hunt group is busy, i then check the channels and there are open calls that were hung up long ago. i thought it was a zap problem but then i saw the same problem with iax2 calls. its becoming a huge issue because if i dont reboot asterisk several times a day, all our lines get filled up with dead calls. I am now running 1.2.21.1 asterisk with the same problem. Please help. Mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slow list
Hi, Am Donnerstag, den 05.07.2007, 14:58 -0400 schrieb Andrew Kohlsmith: On Thursday 05 July 2007 2:38 pm, Doug Lytle wrote: Already did that. I use ASSP for filtering. Digium and associated mailing lists are white listed. There was only 1 attempt for deliver and there were no delays. I subscribe to 10 mailing lists (Including the dev list) and they are not having issues. By the way, the only reason I'm able to respond to your messages and I'm watching the archives at lists.digium.com I am having no issues with Digium's lists. They get a little laggy at times, but generally are fast enough. I don't agree. Your mail was delivered on 12 July (7 days delay): Received: from lists.digium.com (EHLO lists.digium.com) [216.207.245.17] by mx0.gmx.net (mx083) with SMTP; 12 Jul 2007 18:52:37 +0200 Received: from localhost ([127.0.0.1] helo=INXS.digium.internal) by lists.digium.com with esmtp (Exim 4.63) (envelope-from [EMAIL PROTECTED]) id 1I6Wa3-0006CM-7P; Thu, 05 Jul 2007 14:01:31 -0500 I think, something is wrong. Currently the most recent mail I got from the list is from 11 July! Regards Karsten ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distribution lists for voicemail
On Fri, 2007-07-13 at 08:20 -0500, Jadrien Wauthier wrote: Does anyone know of a good way to leave one voicemail message, and the message be forwarded to multiple voicemail boxes at once. You can pass multiple mailboxes to the VoiceMail() dialplan application separated by ampersands, like this: exten = 123,1,Voicemail([EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED]) As I understand it, the greetings will come from the first mailbox specified, but a copy of the voicemail will be left in all of the specified mailboxes. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-addons compilation error: dereferencing pointer to incomplete type
On Fri, Jul 13, 2007 at 10:06:13PM +0800, Jeremy Malcolm wrote: I am having trouble getting asterisk-addons 1.4.2 to compile (after a successful configure). Asterisk itself (and AsteriskGUI) compile fine. Please provide the following: * Version of Asterisk you have installed * Linux distribution and version * MySQL is installed from a distro package, right? I get: cdr_addon_mysql.c: In function `handle_cdr_mysql_status': cdr_addon_mysql.c:91: error: dereferencing pointer to incomplete type cdr_addon_mysql.c:93: error: dereferencing pointer to incomplete type -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distribution lists for voicemail
On Fri, 2007-07-13 at 08:20 -0500, Jadrien Wauthier wrote: Hi, I am sure someone has already asked this, but I am fairly new to the list, so I haven't seen anything on this. Does anyone know of a good way to leave one voicemail message, and the message be forwarded to multiple voicemail boxes at once. I realize that we could leave one message in a group box and have the message forwarded to a group email address, but their are some users that want the message in their actual voicemail box. Clearly a case where you have done no research at all because it is well documented. I'll be kind so here it goes: Use the command: Voicemail(12345) Basically separate the mailbox number with an . The greeting will be from the first mailbox you specify. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Media Proxy Mode in Asterik: SIP and H.323
Hi List; All we know that in voice, there are a type of communications between endpoints, for example: in some communications we do a proxy for media and signaling while other communications we do a proxy for only signaling. Where I can determine these things in Asterisk if I am using SIP and if I am using H.323? Regards -- IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 00965 9849460 Luggage? GPS? Comic books? Check out fitting gifts for grads at Yahoo! Search http://search.yahoo.com/search?fr=oni_on_mailp=graduation+giftscs=bz ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-addons compilation error: dereferencing pointer to incomplete type
Jeremy Malcolm wrote on 7/13/07 10:06 AM: I am having trouble getting asterisk-addons 1.4.2 to compile (after a successful configure). Asterisk itself (and AsteriskGUI) compile fine. I get: [snipped] Sounds like you're missing the -devel package for MySQL would be my first guess. -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.org/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Macro: s-NOANSWER, _s-.
On Fri, 2007-07-13 at 09:16 -0700, bilal ghayyad wrote: I have this example for Macro and I am not able to understand some line, if any one can help me plz :)- [snip] exten = _s-.,1,Goto(s-NOANSWER,1) Also, what does it mean _s-. ? It indicated for which dialing number? The underscore signifies that it's a pattern match, and the period means one more more characters after the s and the dash. In other words, it's a patter that would match s-CONGESTION or anything else that started with an s and a dash. -Jared ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Macro: s-NOANSWER, _s-.
Hi List;
I have this example for Macro and I am not able to
understand some line, if any one can help me plz :)-
[macro-voicemail]
exten = s,1,Dial(${ARG1},20)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})
exten = s-NOANSWER,2,Goto(incoming,s,1)
exten = s-BUSY,1,Voicemail(b${MACRO_EXTEN})
exten = s-BUSY,2,Goto(incoming,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
When it will go for the line:
exten = s-NOANSWER,2,Goto(incoming,s,1)
And when it will go for the line:
exten = s-BUSY,2,Goto(incoming,s,1)
And when it will go for the line:
exten = _s-.,1,Goto(s-NOANSWER,1)
Also, what does it mean _s-. ? It indicated for which
dialing number?
Any help?
Regards
--
IP Telephony and Contact Center Engineer
Eng. Bilal Ghayad
Mobile: 00965 9849460
Park yourself in front of a world of choices in alternative vehicles. Visit the
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Re: [asterisk-users] Media Proxy Mode in Asterik: SIP and H.323
Bilal, Asterisk is an IP PBX and thus a back-to-back user agent; by default, it will proxy media. The only way to disengage it from the media stream is to use signaling protocol-specific mechanisms to coax the endpoints into talking to each other directly; in SIP, this can be done via re-INVITEs a la the canreinvite= option for SIP peers in sip.conf. The H.323 stack in Asterisk may or may not have a similar option, and it may or may not be compatible with SIP on a signaling level. That's if you're connecting one endpoint that's H.323 and one that's SIP. I imagine there's probably a way to do media stream handoff between two legs that are natively H.323 on both ends. But that's the determinant. Also, remember that even if you hand off the media, Asterisk still stays in the signaling path. This is all the more true if you're using it as a signaling gateway between heterogenous protocols. -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] limit simultaneous calls
On Fri, 13 Jul 2007, Mark Quitoriano wrote: is there a way to limit an account to do simultaneous calls in sip and iax? Among other solutions that have been proposed, you can always use global variables or AstDB values to keep per-peer reference counts and increment and decrement them as calls get set up and torn down, and enforce limits that way. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH stop and resume when i hold
I have a strange comportment of the MOH system on my asterisk. When i respond to a call and after fews second i set this call in hold mode the correspondent listen the music fine. When i re-take my correspondent at T0 instant the music is paused. And when i re-hold him at T60 (60 second later) the sound is always at T0 when he was stopped at T0. So the music is stopped and don't continue (normal comportment) when i speak with peoples. Someone know the problem? a solution? What you're describing is the asterisk native files method for MOH. This behavior is the way it was designed. If you want continuous MOH, I'd suggest using a third-party player like madplay or mpg123. You can read more here: http://www.voip-info.org/wiki/index.php?page=Asterisk%20config%20musiconhold.conf (Hint: Scroll down to the section labeled Pre-Asterisk 1.2 Information Below) If you are really using mpg123 or madplay, it should not work that way. Can you post your musiconhold.conf file? - Noah ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can Asterisk hear on two IP addresses? And can I do routing for calls from private to public or public to private IP addresses
Hi List; Can asterisk hear (receive) calls on two IP addresses? How? If yes, then: If I have a VPN router, and my Asterisk server connected to two network cards, one has a private IP address (192.168.0.2) connected to the VPN router (192.168.0.1) and another network card has a private IP address (193.111.196.249) connected directly to the outside default gateway (193.111.196.240), where the VPN default gateway for outside is also (193.111.196.240), then: If I received a call on the network card of IP: 192.168.0.2 then can I route the call for another softswitch server has a public IP address (in another county and another network)? If yes, then is there some condition on this kind of call routing (for example: the communication mode to be full proxy for media and signaling or it can be a proxy only for signaling)? Any help? Regards --- IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 0965 9849460 Be a PS3 game guru. Get your game face on with the latest PS3 news and previews at Yahoo! Games. http://videogames.yahoo.com/platform?platform=120121 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfer Question
Hi Mark - I'm having a tough time figuring out how to do something. If I have an operator (which could potentially be in their own context) and an internal-only context, is it possible to make it so the operator can call the internal-only context but *NOT* transfer calls to it? Sort of. You can create a special extension in the operator's context with a Goto() statement. Something like this: [operator] exten = 100,1,Goto(internal,prompt,1) Then in the internal context: [internal] exten = prompt,1,Background(who-do-you-want-to-call) exten = prompt,2,Waitexten(10) So, when the operator dials 100, he/she can then dial an extension in the internal context. Normal transfer from [operator] to [internal] would not be allowed. - Noah ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH stop and resume when i hold
2007/7/13, Noah Miller [EMAIL PROTECTED]: I have a strange comportment of the MOH system on my asterisk. When i respond to a call and after fews second i set this call in hold mode the correspondent listen the music fine. When i re-take my correspondent at T0 instant the music is paused. And when i re-hold him at T60 (60 second later) the sound is always at T0 when he was stopped at T0. So the music is stopped and don't continue (normal comportment) when i speak with peoples. Someone know the problem? a solution? What you're describing is the asterisk native files method for MOH. This behavior is the way it was designed. If you want continuous MOH, I'd suggest using a third-party player like madplay or mpg123. You can read more here: http://www.voip-info.org/wiki/index.php?page=Asterisk%20config%20musiconhold.conf (Hint: Scroll down to the section labeled Pre-Asterisk 1.2 Information Below) If you are really using mpg123 or madplay, it should not work that way. Can you post your musiconhold.conf file? - Noah Yes i know. Some weeks ago it was functioning perfectly but since some days it was wrong. My moh.conf (for mpg123): [default] mode=quietmp3 directory=/usr/share/asterisk/mohmp3 random=yes And if i want use the integrated mp3 support of asterisk : [default] mode=files directory=/usr/share/asterisk/mohmp3 random=yes For me it's more a usage of the mpg123 that asterisk pause or a timer problem. Because my configuration functioned before. I don't know if this is a kernel update (i use the default kernel of debian and disable acpi for the zdummy driver) could do this kind of problem. So when i hang-up again a call and recall my extension with musiconhold it restart at the moment when it was stopped with mpg123 / madplay and at the beginning of a random sound with native mp3. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trials with 1.4
At 12:20 PM 7/12/2007, you wrote: Ira wrote: I decided to try 1.4 today. Zaptel builds, installs and seems to load correctly and Asterisk builds, installs and works but no Zap lines. When I typed make menuselect for Asterisk and went poking around I discovered that chan_zap is marked XXX and I can't figure out why that might be. Any suggestions? Make sure you re-run the configure script for Asterisk after installing the 1.4 version of zaptel. So I need to build and install Zaptel 1.4 before the configure script will allow me to build Zap support into Asterisk? Seems a bit draconian, but whatever. Thanks for the hint, I'll try it and see how it goes. Ira ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can Asterisk hear on two IP addresses? And can I do routing for calls from private to public or public to private IP addresses
Bilal, There is no technical difference, from Asterisk's point of view, between bridging call legs from two different subnets that have local interfaces versus bridging call legs from two foreign IP destinations. As long as they are routable and reachable, they can be connected. So, I think the short answer to your question is yes, provided I'm understanding it correctly. Thanks, -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trials with 1.4
On Fri, Jul 13, 2007 at 09:58:30AM -0700, Ira wrote: At 12:20 PM 7/12/2007, you wrote: Ira wrote: I decided to try 1.4 today. Zaptel builds, installs and seems to load correctly and Asterisk builds, installs and works but no Zap lines. When I typed make menuselect for Asterisk and went poking around I discovered that chan_zap is marked XXX and I can't figure out why that might be. Any suggestions? Make sure you re-run the configure script for Asterisk after installing the 1.4 version of zaptel. So I need to build and install Zaptel 1.4 before the configure script will allow me to build Zap support into Asterisk? The configure script tries to detect Zaptel. Frankly all it needs is zaptel.h (or actually: the Zaptel kernel/userspace API). It also needs libtonezone. Unrelated: BadWorkaround If you really must build Asterisk's chan_zap vs. a copy of Zaptel that is not installed on your system, but rather an extracted zaptel source tree located at the path ZAPTEL_PATH (e.g: ZAPTEL_PATH=/usr/src/zaptel-trunk ), then use the following: ln -s . $ZAPTEL_PATH/include ln -s . $ZAPTEL_PATH/zaptel ./configure --with-zaptel=$ZAPTEL_PATH --with-zaptel-vldtmf=$ZAPTEL_PATH \ --with-tonezone=$ZAPTEL_PATH You will need a built tonezone in the Zaptel directory. As you know what you're doing, you can assume it is identical enough to the copy of libtonezone that is already installed on your system (or will be, later on). /BadWorkaround -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfer Question
Noah Miller wrote: Sort of. You can create a special extension in the operator's context with a Goto() statement. Something like this: [operator] exten = 100,1,Goto(internal,prompt,1) Then in the internal context: [internal] exten = prompt,1,Background(who-do-you-want-to-call) exten = prompt,2,Waitexten(10) So, when the operator dials 100, he/she can then dial an extension in the internal context. Normal transfer from [operator] to [internal] would not be allowed. - Noah This might work, but I don't want people to have to remember to dial 100 if they need to call a certain set of extensions. I know that the internal numbers all have 3 digits in their caller-id. Maybe have a different action if the caller-id is not exactly 3 digits? Mark ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Macro: s-NOANSWER, _s-.
Dear Jared; Thanks for your kindly help. But what do u mean by more characters? What that pattern that will contain a character? Also, what that pattern that will contain a dash (-)? Regarding to the s-CONGESTION then what it means by CONGESTION word? Why u used here CONGESTION? Regards, - IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 00965 9849460 I have this example for Macro and I am not able to understand some line, if any one can help me plz :)- [snip] exten = _s-.,1,Goto(s-NOANSWER,1) Also, what does it mean _s-. ? It indicated for which dialing number? The underscore signifies that it's a pattern match, and the period means one more more characters after the s and the dash. In other words, it's a patter that would match s-CONGESTION or anything else that started with an s and a dash. Regards Bilal Ghayad Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news, photos more. http://mobile.yahoo.com/go?refer=1GNXIC ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Selling a Digium TDM400P w/ 4 FXO cards
Sorry in advance if this is slightly off topic. I'm selling a Digium TDM400P with 4 FXO ports for $300 plus $20 shipping costs. This is an authentic Digium TDM400P. It's brand-new and was only installed into a system once and then removed (we decided to go with T1 spans instead). Please email me off list if you're interested in purchasing it. Thanks! -- Kenneth Shaw ExpiTrans, Inc. 129 W. Wilson St., Suite 204 Costa Mesa, CA 92627 tel: 949.650.4600 fax: 949.642.6044 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] b410p and DTMF: dtmfthreshold in 1.2.18 zaptel drivers please?
I have the same problem as the following: http://threebit.net/mail-archive/asterisk-users/msg36602.html I am using a b410p (Wildcard) in TE mode on 4 channels of ISDN2e and it totally fails to detect incoming DTMF. The newer mISDN drivers have a parameter dtmfthreshold where you can tune the volume above which DTMF detection is initiated. However (I've tried it) the newest (1.1.14/15) mISDN drivers don't work with the b410p (the second port will not come up and Asterisk refuses to route inbound or outbound). There is a define DMTF_THRESH in the zaptel sources but I think that only applies to Zap-type cards - I tried adjusting this in the source from the default of 4000 down to as low as 400 and still no joy. In the mISDN.org sources dtmfthresold is a module parameter and appears in various dsp_* source files - but not at all in the Digium sources. Anyone got inbound DTMF working with this card yet? I am using the 1.2.18 release of the zaptel drivers from Digium - I presume the 1.4.x are only meant for asterisk 1.4.x and wouldn't want to break my PBX by trying them (I'm running ast 1.2.20). This is pretty high priority for me as we need to add an IVR menu - without DTMF it's obviously going to be a futile exercise. Cheers Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. Transact is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 1200 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Media Proxy Mode in Asterik: SIP and
Dear Alex; Thanks for your kindly reply. Please explain for me what do u mean exactly in a la in the following sentence u wrote it below? in SIP, this can be done via re-INVITEs a la the canreinvite= option for SIP peers in sip.conf Another thing, do u mean that it is easier (better) if we need H.323 endpoint to talk with SIP endpoint then we use full proxy (signaling + media) so that will solve miss compatibility issues? By the way: Asterisk allow H.323 to talk with SIP? Regards, --- IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 00965 9849460 Bilal, Asterisk is an IP PBX and thus a back-to-back user agent; by default, it will proxy media. The only way to disengage it from the media stream is to use signaling protocol-specific mechanisms to coax the endpoints into talking to each other directly; in SIP, this can be done via re-INVITEs a la the canreinvite= option for SIP peers in sip.conf. The H.323 stack in Asterisk may or may not have a similar option, and it may or may not be compatible with SIP on a signaling level. That's if you're connecting one endpoint that's H.323 and one that's SIP. I imagine there's probably a way to do media stream handoff between two legs that are natively H.323 on both ends. But that's the determinant. Also, remember that even if you hand off the media, Asterisk still stays in the signaling path. This is all the more true if you're using it as a signaling gateway between heterogenous protocols. -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news, photos more. http://mobile.yahoo.com/go?refer=1GNXIC ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Media Proxy Mode in Asterik: SIP and
Bilal, On Fri, 13 Jul 2007, bilal ghayyad wrote: Please explain for me what do u mean exactly in a la in the following sentence u wrote it below? in SIP, this can be done via re-INVITEs a la the canreinvite= option for SIP peers in sip.conf It is an English colloquialism that enjoys wide currency; stolen from French, where it means in the manner of or in the style of, for instance, a la provençale (country style). So, in this case I mean that re-INVITEs are accomplished _by way of_ the 'canreinvite' option. Despite the nomenclature, 'canreinvite' in sip.conf actually means not merely CAN attempt to re-INVITE but WILL attempt to re-INVITE as a point of default behaviour. Another thing, do u mean that it is easier (better) if we need H.323 endpoint to talk with SIP endpoint then we use full proxy (signaling + media) so that will solve miss compatibility issues? Well, an H.323 endpoint cannot speak directly to a SIP endpoint without a signaling gateway to convert the messages. If you are referring purely to passing RTP media, I suppose it does not matter, if you can broker that handoff somehow between H.323 and SIP. I am not sure if Asterisk has this capability. By the way: Asterisk allow H.323 to talk with SIP? In principle, yes. Asterisk can act as an H.323 gatekeeper, although not as an endpoint (which is irrelevant to your purpose anyway). By far the easiest and cleanest solution is to run both the signaling and the media through Asterisk, if you're making a SIP - H.323 call. This is the approach least likely to cause any compatibility issues since Asterisk is intermediating in the transaction, and I am not sure if an alternative is even possible. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trials with 1.4
Ira wrote: So I need to build and install Zaptel 1.4 before the configure script will allow me to build Zap support into Asterisk? Seems a bit draconian, but whatever. Thanks for the hint, I'll try it and see how it goes. Well, what would you expect? Should the configure script say I can't find zaptel, but I *think* this system will have zaptel installed at some point before Asterisk gets compiled? The whole point of the configure script is to check the system for what is installed, to see what parts of Asterisk can be built. -- Russell Bryant Software Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trials with 1.4
At 01:00 PM 7/13/2007, you wrote: Ira wrote: So I need to build and install Zaptel 1.4 before the configure script will allow me to build Zap support into Asterisk? Seems a bit draconian, but whatever. Thanks for the hint, I'll try it and see how it goes. Well, what would you expect? Should the configure script say I can't find zaptel, but I *think* this system will have zaptel installed at some point before Asterisk gets compiled? The whole point of the configure script is to check the system for what is installed, to see what parts of Asterisk can be built. Well, in that case, having the dependency for Zaptel indicate you need to build Zaptel and then run Configure would be useful instead of including a list of dependencies running off the edge of the screen that I can't read. In my case, Zaptel 1.4 was installed and running and I thought I ran configure again, but maybe not. What's the downside of just saying, This is useless as Zaptel does not seem to be installed? In 1.2 Zaptel was always built or maybe I just did it right by accident the first time. I'm not stupid, and I'm sort of a Linux newbie, but it seems you're making 1.4 harder to install that 1.2 and that seems a step backwards. Is there a problem if I choose to build Zaptel support on a machine without Zap installed? Ira ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Different SIP From and Auth?
Looks like this isn't possible. I wonder if there's a bug open on this? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Ma Sent: Thursday, July 12, 2007 1:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Different SIP From and Auth? Hi I have asked this questions,but have no answer :) I also want Asterisk do not check to head with digest username in registration,how can we do that? On 7/12/07, Douglas Garstang [EMAIL PROTECTED] wrote: Is it possible to have Asterisk allow the From address in a SIP invite to be different to the required digest username? The auth parameter supposedly allows it, but whether or not I set auth to be what the UA sends as the digest username, Asterisk just complains that the from and the digest are different, and it gives up. Doug ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trials with 1.4
On Fri, Jul 13, 2007 at 01:42:06PM -0700, Ira wrote: At 01:00 PM 7/13/2007, you wrote: Ira wrote: So I need to build and install Zaptel 1.4 before the configure script will allow me to build Zap support into Asterisk? Seems a bit draconian, but whatever. Thanks for the hint, I'll try it and see how it goes. Well, what would you expect? Should the configure script say I can't find zaptel, but I *think* this system will have zaptel installed at some point before Asterisk gets compiled? The whole point of the configure script is to check the system for what is installed, to see what parts of Asterisk can be built. Well, in that case, having the dependency for Zaptel indicate you need to build Zaptel and then run Configure would be useful instead of including a list of dependencies running off the edge of the screen that I can't read. In my case, Zaptel 1.4 was installed and running and I thought I ran configure again, but maybe not. What's the downside of just saying, This is useless as Zaptel does not seem to be installed? What version of Zaptel do you have installed? In 1.2 Zaptel was always built or maybe I just did it right by accident the first time. I'm not stupid, and I'm sort of a Linux newbie, but it seems you're making 1.4 harder to install that 1.2 and that seems a step backwards. Is there a problem if I choose to build Zaptel support on a machine without Zap installed? In 1.2 there was actually some logic in channels/Makefile as to whthether or not to build some channels. Which means that if you wanted to customize pathes or something you had to edit those makefiles. Now you can pass the path to autoconf (just like with most other projects). -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trials with 1.4
Ira wrote: Well, in that case, having the dependency for Zaptel indicate you need to build Zaptel and then run Configure would be useful instead of including a list of dependencies running off the edge of the screen that I can't read. In my case, Zaptel 1.4 was installed and running and I thought I ran configure again, but maybe not. What's the downside of just saying, This is useless as Zaptel does not seem to be installed? In 1.2 Zaptel was always built or maybe I just did it right by accident the first time. I'm not stupid, and I'm sort of a Linux newbie, but it seems you're making 1.4 harder to install that 1.2 and that seems a step backwards. Is there a problem if I choose to build Zaptel support on a machine without Zap installed? Well, it was certainly the intent to make things easier. Also, it isn't possible to build Zaptel support in Asterisk without Zaptel installed. Installing zaptel installs the zaptel.h header file, which is needed for compiling most of the zaptel related code. That was true for 1.2 as well. In 1.4, this check is done by the configure script instead of by the Makefile. However, using the configure script to do these checks provides some benefits. Let me run through a list of some of the benefits of the 1.4 system of building Asterisk. * In Asterisk 1.2, the Makefile had hard coded paths to look for header files to determine whether a library was installed. In Asterisk 1.4, you can pass a path as an option to the configure script so that you can build against libraries that are in non-standard locations. * The 1.2 header file check did not guarantee that what is installed is actually going to be able to be used by Asterisk. The configure script allows us to build a test program against each library to ensure that it is actually usable, and that it is a compatible version. So, in 1.2 where it was possible that a module would fail to build due to library incompatibilities, this will not happen with 1.4. * In Asterisk 1.2, the Makefiles would build every module it could, without any way to control that. In Asterisk 1.4, there is a console menu interface for choosing which modules get built (make menuselect). Furthermore, if modules can not be built due to an unmet dependency, the menuselect interface gives you an easy way to see that, as well as find out what libraries that module depends on. Selections through this interface are stored in a file (menuselect.makeopts), which can be copied to other systems, or placed in /etc/asterisk.makeopts or ~/.asterisk.makeopts to ensure the same build configuration for multiple builds. I am hoping that these added features justify the fact that you have to explicitly re-run the configure script to go check for newly installed dependencies on the system. -- Russell Bryant Software Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google acquires Grand Central
Sorry for the very delayed response but wanted to add - the older I get the more I realize that it very often has little to do with Technology. Packaging something into an easy to understand easy to market 'product' is more often than not the key to successful sales. GrandCentral have won business that would never have moved to an asterisk server because even Trixbox is still too complicated. We really need some more off the shelf asterisk appliances that are self contained and plug and play for people that aren't geeks. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Monday, 9 July 2007 1:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google acquires Grand Central On Mon, 9 Jul 2007, Wendell Hamilton wrote: GrandCentral doesn't do anything you can't do with asterisk. What it does do is put those features within reach of an average person by providing a superb user interface for the end user, which allows them to self-administer all of these wonderful features. Indeed, the success of their accomplishment -- as with most aspects of successful business -- seems to have been in effectively *productising* these solutions, and developing viable business processes and workflow in order to make them viably scale to the masses of end-users. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Macro: s-NOANSWER, _s-.
After the Dial application completes, the variable ${DIALSTATUS} will
contain something like BUSY or NOANSWER or CONGESTION (CONGESTION here
means like no free phone lines or no route to destination for example)
Then, immediately after the Dial line is the Goto line
Goto(s-${DIALSTATUS},1)
this will in reality evaluate to something like
Goto(s-CONGESTION,1)
or
Goto(s-BUSY,1)
based on the contents of the DIALSTATUS variable (which is dependent on
the results of the Dial application)
Therefore, you need the extensions s-BUSY and s-NOANSWER etc to deal
with these statuses.
Moj
bilal ghayyad wrote:
Dear Jared;
Thanks for your kindly help.
But what do u mean by more characters? What that
pattern that will contain a character?
Also, what that pattern that will contain a dash (-)?
Regarding to the s-CONGESTION then what it means by
CONGESTION word? Why u used here CONGESTION?
Regards,
-
IP Telephony and Contact Center Engineer
Eng. Bilal Ghayad
Mobile: 00965 9849460
I have this example for Macro and I am not able to
understand some line, if any one can help me plz :)-
[snip]
exten = _s-.,1,Goto(s-NOANSWER,1)
Also, what does it mean _s-. ? It indicated for
which
dialing number?
The underscore signifies that it's a pattern match,
and the period
means
one more more characters after the s and the dash. In
other words,
it's
a patter that would match s-CONGESTION or anything
else that started
with an s and a dash.
Regards
Bilal Ghayad
Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail,
news, photos more.
http://mobile.yahoo.com/go?refer=1GNXIC
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Re: [asterisk-users] asterisk-addons compilation error: dereferencing pointer to incomplete type
On Fri, Jul 13, 2007 at 12:00:11PM -0500, [EMAIL PROTECTED] wrote:
I am having trouble getting asterisk-addons 1.4.2 to compile (after a
successful configure). Asterisk itself (and AsteriskGUI) compile fine.
Please provide the following:
* Version of Asterisk you have installed
* Linux distribution and version
* MySQL is installed from a distro package, right?
1.4.5 compiled from source on Debian sarge (with backported packages
from etch as needed by Asterisk) and MySQL 5.0.32.
Sounds like you're missing the -devel package for MySQL would be my
first guess.
I have libmysqlclient15-dev, which should be correct...?
Thanks.
--
Jeremy Malcolm LLB (Hons) B Com
Internet and Open Source lawyer, IT consultant, actor
host -t NAPTR 1.0.8.0.3.1.2.9.8.1.6.e164.org|awk -F! '{print $3}'
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[asterisk-users] Info about Providers
To everyone on the list I put a site on line the URL is *http://bochterservices.com/phpbb/ *This is for any information on Good or Bad ITSP You can post any problems you had with the provider You can Vote on the provider This is for allowing multiple viewpoints to be heard. If a provider receives a bad review, they are more than welcome to post So long as the exchange is fairly open and truthful And this list will be carefully moderated Please do some posting! By the way I am looking for moderators for the list if you want to help let me know. -- Best regards, Al Bochter http://www.BochterServices.com --- See what we are selling at auction http://www.epier.com/auctions.asp?bochterservices --- Take a look at our online store http://www.bochterservices.com/onlinestore/ --- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Micros-Fidelio - billing in hotel
Chris Mason (Lists) wrote: Lee Jenkins wrote: I'd say that Micro is the MS of Restaurant POS. We replace their systems regularly ;) I'm curious what with? www.datatrakpos.com Notice that I didn't say en masse but yes, we do replace a few Micros systems a year. Same thing with some of the other brands out there. We've had a few switched from us over the years as well. Basic attrition, I guess. -- Warm Regards, Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Micros-Fidelio - billing in hotel
Tomislav Parcina wrote: There is hotel application weary popular in Croatia - Micros-Fidelio. Now I need to connect Asterisk with this application for purpose of billing. Thing is that hotel would like to give customer one bill for every service that he used while he was in hotel. Has anybody connected Asterisk with Micros-Fidelio? As I understand this isn't some local developed application, it's something that is used world wide. Any informations are welcome. I wrote a middleware bridge (TCP = Serial) for Micros a 2 or 3 years back and it was relatively simple. This was the serial interface for the 8700 standard. If I remember correctly, it was a simple string that was broken up into fixed length fields like char 1 through 10 was a field and chars 11 through 15 was a field, etc. If you need help, email me off list and I'll look for that source code. Lucky for you it was written in pascal so its easy to read ;) Warm Regards, Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] calling from ACT
I was wondering if any of you guys are aware of ability to call customers by click on customer's phone number in ACT? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trials with 1.4
At 02:16 PM 7/13/2007, you wrote: What version of Zaptel do you have installed? I discovered the issue when I rebooted the machine and the latest version of Zap 1.4 installed and ran. In 1.2 Zaptel was always built or maybe I just did it right by accident the first time. I'm not stupid, and I'm sort of a Linux newbie, but it seems you're making 1.4 harder to install that 1.2 and that seems a step backwards. Is there a problem if I choose to build Zaptel support on a machine without Zap installed? In 1.2 there was actually some logic in channels/Makefile as to whthether or not to build some channels. Which means that if you wanted to customize pathes or something you had to edit those makefiles. Now you can pass the path to autoconf (just like with most other projects). And this is useful to me how? Not to be stupid, but in my world all I know how to do is the default install. I'm not going to think it's not useful for many people, but it doesn't matter to me. In the end my issue would seem to be I did something out of order, though I thought I did it right, and as much as I tried the software only gave meaningless to me messages. I didn't understand and I don't think I've seen it said before that I should run .\configure every time I do anything. I didn't understand what it did and had no idea it was how I'd get Zaptel support to build. Ira ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] open source screen pop software for asterisk
Hi! I am new here. Well I'm doing a call center using asterisk and I'm looking for an open source screen pop software to pop the caller's information, its call history and others things. i was looking around and find the U-rang2 the problem is that it isn't open source. if someone knows about an open source screen pop please tell me. thanks in advance renzzo ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfer Call to Cell Phone
This email set a record--taking 12 day to reach me. If it's something in my control, I'd love to know what so I can fix it. Return-Path: [EMAIL PROTECTED] Delivered-To: [EMAIL PROTECTED] Received: (qmail 21508 invoked from network); 13 Jul 2007 23:45:38 - Received: from lists.digium.com (216.207.245.17) by lakeminnetonkaexcelsiorrotaryclub.org with (AES256-SHA encrypted) SMTP; 13 Jul 2007 23:45:38 - Received: from localhost ([127.0.0.1] helo=INXS.digium.internal) by lists.digium.com with esmtp (Exim 4.63) (envelope-from [EMAIL PROTECTED]) id 1I5CDT-0001O3-Dw; Sun, 01 Jul 2007 22:04:43 -0500 Received: from exprod8mx64.postini.com ([64.18.3.164] helo=psmtp.com) by lists.digium.com with smtp (Exim 4.63) (envelope-from [EMAIL PROTECTED]) id 1I5CDL-0001No-LD for asterisk-users@lists.digium.com; Sun, 01 Jul 2007 22:04:35 -0500 Received: from source ([208.65.112.50]) by exprod8mx64.postini.com ([64.18.7.10]) with SMTP; Sun, 01 Jul 2007 20:04:34 PDT Received: from ([127.0.0.1]) with MailEnable ESMTP; Sun, 01 Jul 2007 22:04:33 -0500 Message-ID: [EMAIL PROTECTED] Date: Sun, 01 Jul 2007 22:04:32 -0500 From: OCOSA ListAcct [EMAIL PROTECTED] User-Agent: Thunderbird 2.0.0.4 (Windows/20070604) MIME-Version: 1.0 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com References: [EMAIL PROTECTED] In-Reply-To: [EMAIL PROTECTED] X-pstn-neptune: 0/0/0.00/0 X-pstn-levels: (S:99.9/99.9 FC:95.5390 LC:95.5390 R:95.9108 P:95.9108 M:97.0282 C:98.6951 ) X-pstn-settings: 3 (1.:1.) s fc lc gt3 gt2 gt1 r p m c X-pstn-addresses: from [EMAIL PROTECTED] [db-null] Subject: Re: [asterisk-users] Transfer Call to Cell Phone X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.9 Precedence: list Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com List-Id: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users, mailto:[EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:asterisk-users@lists.digium.com List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users, mailto:[EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Sender: [EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] --Don Don Kelly PCF Corp Real Support for your Virtual Office 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of OCOSA ListAcct Sent: Sunday, July 01, 2007 10:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Transfer Call to Cell Phone works perfectThanks -- Otis John Faubion wrote: We do have full features on our lines so both lines are free once the transfer is complete. We also have toll calls on our lines so it would not be a problem, so I do not have to worry about ATT dropping the The issue really isn't whether you have the ability to make toll calls on your line. The concern here is in what the regulatory agencies call toll bridging which is using a system to relay a call from one local calling are to another local calling area to avoid a toll charge. This is one of those gray areas that can become a problem if your not careful. The problem comes up if you have customers that can call you as a local call and you are forwarding them on to another party that is a local call for you but would be a toll call for the customer. This is essentially what toll bridging is about. Now your not likely to have to worry about the legal ramifications of this since your merely connecting the customer with an extension of your company, namely your salesman. Where this could become a problem for you would be in transferring the customer using the same pots line. The reason is that ATT is handling the transfer. When you transfer the call, it essentially becomes a new call. The main difference is that you have provided the called number. So the software in the Class 5 (End office) switch, takes the number you provide and runs the call through its routing translations (similar to the Asterisk dialing plan) and if it determines that the destination number is outside the originators Local Area Transport Area or LATA, then it will either drop the originator to a message that says, You must first dial a 0 or 1 before calling this number or it may deny the transfer allowing you to stay connected to the customer. Neither one looks very professional. The only way around this would be to provide another line or trunk to pass the call down. Now if your not in an overlapping LATA this probably isn't an issue. The only way I can
