[asterisk-users] Bad hangup event cause
Hello, i have a problem with the hangup cause received from the AMI in the Hangup events. All calls that arent answered after ringing are returning hangup cause 16 (normal clearing) instead 19. Im running asterisk 1.4.11, the calls are generated to a SIP peer using the AMI originate command. This is the 'sip debug' output: Reliably Transmitting (no NAT) to 192.168.0.70:5060: INVITE sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK286113b7;rport From: 123 sip:[EMAIL PROTECTED];tag=as0cd1aab0 To: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 27 Aug 2007 05:53:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 238 v=0 o=root 21676 21676 IN IP4 192.168.0.1 s=session c=IN IP4 192.168.0.1 t=0 0 m=audio 15274 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- gw*CLI --- SIP read from 192.168.0.70:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK286113b7;rport From: 123 sip:[EMAIL PROTECTED];tag=as0cd1aab0 To: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp;tag=2035093099 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Server: Cisco ATA 186 v3.2.1 atasip (050616A) Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Content-Length: 0 - --- (9 headers 0 lines) --- gw*CLI --- SIP read from 192.168.0.70:5060 --- SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK286113b7;rport From: 123 sip:[EMAIL PROTECTED];tag=as0cd1aab0 To: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp;tag=2035093099 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Contact: 1 sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp Server: Cisco ATA 186 v3.2.1 atasip (050616A) Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Content-Length: 0 - --- (10 headers 0 lines) --- Reliably Transmitting (no NAT) to 192.168.0.70:5060: OPTIONS sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK49966ec7;rport From: asterisk sip:[EMAIL PROTECTED];tag=as0916f4ed To: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 27 Aug 2007 05:53:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- gw*CLI --- SIP read from 192.168.0.70:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK49966ec7;rport From: asterisk sip:[EMAIL PROTECTED];tag=as0916f4ed To: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp;tag=3724167432 Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS Server: Cisco ATA 186 v3.2.1 atasip (050616A) Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces Content-Length: 250 Content-Type: application/sdp v=0 o=2 19680158 19680158 IN IP4 192.168.0.70 s=ATA186 Call c=IN IP4 192.168.0.70 t=0 0 m=audio 16386 RTP/AVP 0 8 4 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:4 G723/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 - --- (11 headers 11 lines) --- Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS Reliably Transmitting (no NAT) to 192.168.0.70:5060: OPTIONS sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK02be2790;rport From: asterisk sip:[EMAIL PROTECTED];tag=as6ba5f9aa To: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 27 Aug 2007 05:53:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- gw*CLI --- SIP read from 192.168.0.70:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK02be2790;rport From: asterisk sip:[EMAIL PROTECTED];tag=as6ba5f9aa To: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp;tag=2035093099 Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS Server: Cisco ATA 186 v3.2.1 atasip (050616A) Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces Content-Length: 250 Content-Type: application/sdp v=0 o=1 19680166 19680166 IN IP4 192.168.0.70 s=ATA186 Call c=IN IP4 192.168.0.70 t=0 0 m=audio 16384 RTP/AVP 0 8 4 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:4 G723/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 - --- (11 headers 11 lines) --- Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS Scheduling destruction of SIP
[asterisk-users] call forwading problem DTMF
Dear all I have recently install TE120P Digium E1 card now everything is fine and working i have connect my asterisk with avaya but when anybody transfer call from avaya i got this error on my asterisk consol [Aug 27 14:46:50] WARNING[19527]: app_dial.c:741 wait_for_answer: Unable to forward voice or dtmf -- Hungup 'Zap/32-1' I m waiting for your reply Satish Patel - Fussy? Opinionated? Impossible to please? Perfect. Join Yahoo!'s user panel and lay it on us.___ --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] Is it possible to register without sending the password
Dear Philipp; Kindly find the part of the configuration as below: [general] allow=all disallow is comment by ( ; ). [bilal_sip] type=friend context=internal host=dynamic canreinvite=no dtmfmode=rfc2833 So where is the problem? The endpoint does not register and nothing appear on trace level 3. And the amazing thing that if the endpoint send wrong username (for example: bilal_sip100) then it does not register, but we see the failed attempt of registration at Asterisk CLI (with trace level = 3). Please any help? Regards Bilal Ghayad Mobile: 009659849460 If secret enabled, then some endpoints can not register (maybe due to compatibility in reading the negotiation packets), so what is the solution? I'm sure they can. Maybe you could tell the list which endpoints don't work? Also in SIP registration: why I do not see the log for registration packets periodically while I can see this in IAX2? Is it related to my v tracing level? Probably. How about you try with more vvv? If you *really* need to see what's going on you might add verbose and debug to the console= entry in logger.conf. But that's probably not what you want. Last point: I noticed that some endpoints that are not able to register (when secret is required), then I was not able to see any log at the asterisk side while SIP client still not registered. At least, it should display the fail for registeration, why does not display it? Is it related to my v tracing level? Where in the same tracing level, I am able to see the registeration fail if the endpoint sent an wrong username. For example if the context was [bilal_sip] and the endpoint username was bilal_1000 then I see a the message (log) that declare that registeration from bilal_1000 failed (ofcourse because bilal_1000 is not configured while bilal_sip is configured in the sip.conf). Could you send the part of your sip.conf? Sounds like a configuration issue. Regards, Philipp Kempgen Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge to see what's on, when. http://tv.yahoo.com/collections/222 ___ --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] foneBRIDGE2 setup
Excellent! - posted on http://oinko.net/astpligg/story.php?title=Asterisk_clusters_with_a_foneBRIDGE2 Thank you l. In data Sun, 26 Aug 2007 12:56:04 +0200, Vicente Aguilar [EMAIL PROTECTED] ha scritto: Hi I've published my Asterisk/foneBRIDGE2/heartbeat setup: config files, scripts... along with a brief description of the architecture and working of the cluster. It's available here: http://www.bisente.com/blog/2007/08/26/asterisk-cluster-fonebridge2/? lan=english Hope somebody finds it useful. :) Regards -- Home of QueueMetrics - http://queuemetrics.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom firmware download
Thank you David! On 8/26/07, Klaverstyn, David C [EMAIL PROTECTED] wrote: http://www.testforme.com/download/ I'll leave the files there for a few days. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Bosch Sent: Monday, 27 August 2007 4:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom firmware download Hi: Doug wrote: At 13:29 8/25/2007, Al lists wrote: Thats just sad, I got SIP 2.2 from trixbox now, but still we need to have some sort of place at least for ourselves to download this stuff. Looking for boot loader now. Which version? http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip 330_320.html#download http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip 430.html#download It's funny how every time this question gets asked, there's some smart guy (who doesn't use Polycom sets himself) who finds these links. (I'm sincerely thankful for the effort, though.) Only authorized resellers can download the current firmware from those URLs. The only guaranteed way to get the current firmware is to get it from a/your reseller. Posting the firmware packages on a third-party site is a violation of Polycom's EULA. Why do they do this? Because they want to control the sales channel. I don't agree with it, but it's how they operate. If you want a more detailed answer, ask Polycom directly, and I wish you luck. Cheers, -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Chan-capi Fedora 7
Nope, it only has chan_capi. I don't have any experience with AVM Fritz cards so I'm afraid I can't help you with it. I think there is an article on voip-info.org that explains howto use a Fritz card with Asterisk. Regards, Patrick Patrick thanks! I guess my question should have been slightly better framed. Has anyone manged to load the drivers for an AVM Fritz Card (AKA BT Speedway card) in Fedora 7? ___ --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] libmfcr2 is giving compilation errors
hi, I am using debian 4.0 with version 2.6.18-4-686 I have downloaded the required files form site asterisk-1.2.24.tar.gz libmfcr2-0.0.3-1.4.tar.bz2 libsupertone-0.0.2.tar.gz libunicall-0.0.3-1.4.tar.bz2 spandsp-20060903.tar.gz I downloaded and installed the files in the follwing sequence spandsp libsupertone libunicall Till here it is compiling and copying .so library to /usr/local/lib/ libmfcr2-0.0.3 source code is giving a lot of definition error [EMAIL PROTECTED]:/usr/src/libmfcr2-0.0.3_1.4$ make make all-am make[1]: Entering directory `/usr/src/libmfcr2-0.0.3_1.4' if /usr/bin/libtool --tag=CC --mode=compile gcc -DHAVE_CONFIG_H -I. -I. -I. -I/u sr/include/libxml2-g -O2 -MT mfcr2.lo -MD -MP -MF .deps/mfcr2.Tpo -c -o mf cr2.lo mfcr2.c; \ then mv -f .deps/mfcr2.Tpo .deps/mfcr2.Plo; else rm -f .deps/mfcr2. Tpo; exit 1; fi gcc -DHAVE_CONFIG_H -I. -I. -I. -I/usr/include/libxml2 -g -O2 -MT mfcr2.lo -MD -MP -MF .deps/mfcr2.Tpo -c mfcr2.c -fPIC -DPIC -o .libs/mfcr2.o In file included from mfcr2.c:66: mfcr2.h:573: error: expected specifier-qualifier-list before 'r2_mf_tx_state_t' mfcr2.c: In function 'select_active_rxtx': mfcr2.c:444: error: 'mfcr2_signaling_state_t' has no member named 'mf_rx_signal' mfcr2.c:456: error: 'mfcr2_signaling_state_t' has no member named 'mf_rx_signal' mfcr2.c: In function 'set_mf_signal': mfcr2.c:558: error: 'mfcr2_signaling_state_t' has no member named 'mf_tx_signal' mfcr2.c:582: error: 'mfcr2_signaling_state_t' has no member named 'mf_tx_signal' mfcr2.c:587: error: 'mfcr2_signaling_state_t' has no member named 'mf_tx_signal' mfcr2.c:593: error: 'mfcr2_signaling_state_t' has no member named 'mf_tx_signal' mfcr2.c: In function 'mf_tone_on_event': mfcr2.c:1413: error: 'mfcr2_signaling_state_t' has no member named 'mf_rx_signal ' mfcr2.c:1416: error: 'mfcr2_signaling_state_t' has no member named 'mf_rx_signal ' mfcr2.c: In function 'mf_tone_off_event': mfcr2.c:1836: error: 'mfcr2_signaling_state_t' has no member named 'mf_rx_signal ' mfcr2.c:1839: error: 'mfcr2_signaling_state_t' has no member named 'mf_rx_signal ' mfcr2.c:1840: error: 'mfcr2_signaling_state_t' has no member named 'mf_rx_signal ' mfcr2.c:1891: error: 'mfcr2_signaling_state_t' has no member named 'super_tone_t x_state' mfcr2.c:1891: error: 'mfcr2_signaling_state_t' has no member named 'super_tones' mfcr2.c:1908: error: 'mfcr2_signaling_state_t' has no member named 'super_tone_t x_state' mfcr2.c:1908: error: 'mfcr2_signaling_state_t' has no member named 'super_tones' mfcr2.c:1922: error: 'mfcr2_signaling_state_t' has no member named 'super_tone_t x_state' mfcr2.c:1922: error: 'mfcr2_signaling_state_t' has no member named 'super_tones' mfcr2.c:1936: error: 'mfcr2_signaling_state_t' has no member named 'super_tone_t x_state' mfcr2.c:1936: error: 'mfcr2_signaling_state_t' has no member named 'super_tones' mfcr2.c:1950: error: 'mfcr2_signaling_state_t' has no member named 'super_tone_t x_state' mfcr2.c:1950: error: 'mfcr2_signaling_state_t' has no member named 'super_tones' mfcr2.c: In function 'check_event': mfcr2.c:2620: error: 'mfcr2_signaling_state_t' has no member named 'mf_tx_signal ' mfcr2.c:2776: error: 'mfcr2_signaling_state_t' has no member named 'mf_tx_signal ' mfcr2.c:2784: error: 'mfcr2_signaling_state_t' has no member named 'mf_tx_state' mfcr2.c:2788: error: 'mfcr2_signaling_state_t' has no member named 'mf_tx_signal ' mfcr2.c:2792: error: 'mfcr2_signaling_state_t' has no member named 'mf_tx_signal ' mfcr2.c:2796: error: 'mfcr2_signaling_state_t' has no member named 'super_tone_t x_state' mfcr2.c: In function 'load_r2_parameter_set': mfcr2.c:2890: error: 'mfcr2_signaling_state_t' has no member named 'mf_tx_state' mfcr2.c:2890: error: too many arguments to function 'r2_mf_tx_init' mfcr2.c: In function 'drop_call': mfcr2.c:3491: error: 'mfcr2_signaling_state_t' has no member named 'super_tone_t x_state' mfcr2.c:3491: error: 'mfcr2_signaling_state_t' has no member named 'super_tones' mfcr2.c:3547: error: 'mfcr2_signaling_state_t' has no member named 'super_tone_t x_state' mfcr2.c:3547: error: 'mfcr2_signaling_state_t' has no member named 'super_tones' mfcr2.c: In function 'create_new': mfcr2.c:3861: error: 'mfcr2_signaling_state_t' has no member named 'super_tones' mfcr2.c:3863: error: 'mfcr2_signaling_state_t' has no member named 'super_tones' mfcr2.c: At top level: mfcr2.c:4364: fatal error: opening dependency file .deps/mfcr2.Tpo: Permission d enied compilation terminated. make[1]: *** [mfcr2.lo] Error 1 make[1]: Leaving directory `/usr/src/libmfcr2-0.0.3_1.4' make: *** [all] Error 2 Can anybody tell me how to remove these errors I converted .src.rpm file of libmfcr2 to .deb file and installed it. I donot know wether is it is instal;led or not Thanka and regards sanchal
Re: [asterisk-users] No LongDistance for 1 Extension?
Seysan wrote: Hi all, I want to limit the outgoing trunk to certain extensions, so for example 6 extensions can call long distance, but 4 other extensions are not allowed to do so. How can I do it in FreePBX specially? I don't know about Trixbox per say, but normally you would have all the handsets that can make long distance calls in one context and all the ones that can't in another, then use dialplan logic to glue it all together. ___ --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] Authenticating SIP user in LDAP database instead of SIP.conf file
On 26/08/07, Abhishek M S [EMAIL PROTECTED] wrote: Dear Mr Galvin, Gavin ;-) Thank you for the links. Had gone through the bug tracker before though. I was specifically referring to the schema for the driver 'Astirectory' and not the one related to the real time LDAP driver for Open LDAP. It's for any LDAP Compliant Directory Server. In the 'Astirectory' documentation there's a file defining the schema for LDAP which is incomplete. By incomplete I mean the Syntax and few other fields are not defined let alone the schema being a static file. I do understand that for Open LDAP a static file schema should be defined. Not really. in the RealTime driver you can specify which LDAP attributes map to which Asterisk Config settings. The only reason why I preferred Astirectory over the LDAP real time driver was the fact that there is no mapping required for SIP users and peers. OK, maybe I need to go and read more about Astirectory. Regards Abhishek On 8/24/07, Gavin Henry [EMAIL PROTECTED] wrote: Please see the official tracker in the Digium buglist: http://bugs.digium.com/view.php?id=5768 Here are the schemas we did for OpenLDAP: http://bugs.digium.com/file_download.php?file_id=14842type=bug http://bugs.digium.com/file_download.php?file_id=14841type=bug Also, for Novell eDirectory, see: http://forge.voicerd.org/frs/?group_id=7release_id=17 Gavin. -- http://www.suretecsystems.com/services/openldap/ ___ --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 -- http://www.suretecsystems.com/services/openldap/ ___ --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] Authenticating SIP user in LDAP database instead of SIP.conf file
I see it is res_config_ldap. You'd be much better using the latest version in the bug tracker. On 27/08/07, Gavin Henry [EMAIL PROTECTED] wrote: On 26/08/07, Abhishek M S [EMAIL PROTECTED] wrote: Dear Mr Galvin, Gavin ;-) Thank you for the links. Had gone through the bug tracker before though. I was specifically referring to the schema for the driver 'Astirectory' and not the one related to the real time LDAP driver for Open LDAP. It's for any LDAP Compliant Directory Server. In the 'Astirectory' documentation there's a file defining the schema for LDAP which is incomplete. By incomplete I mean the Syntax and few other fields are not defined let alone the schema being a static file. I do understand that for Open LDAP a static file schema should be defined. Not really. in the RealTime driver you can specify which LDAP attributes map to which Asterisk Config settings. The only reason why I preferred Astirectory over the LDAP real time driver was the fact that there is no mapping required for SIP users and peers. OK, maybe I need to go and read more about Astirectory. Regards Abhishek On 8/24/07, Gavin Henry [EMAIL PROTECTED] wrote: Please see the official tracker in the Digium buglist: http://bugs.digium.com/view.php?id=5768 Here are the schemas we did for OpenLDAP: http://bugs.digium.com/file_download.php?file_id=14842type=bug http://bugs.digium.com/file_download.php?file_id=14841type=bug Also, for Novell eDirectory, see: http://forge.voicerd.org/frs/?group_id=7release_id=17 Gavin. -- http://www.suretecsystems.com/services/openldap/ ___ --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 -- http://www.suretecsystems.com/services/openldap/ -- http://www.suretecsystems.com/services/openldap/ ___ --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] Prepaid Billing: A2Billing, AstBill, ASTCC
Hi List; I need to use an prepaid billing system with Asterisk, and I do not know which one is more stable and integrated with Asterisk? A2Billing or AstBill or ASTCC? Also, from where I can download it and ready about its configuration? Regards ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 009659849460 Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=listsid=396545433 ___ --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] AstriCon Tutorials
Hello, Can someone outline what tutorials will be covered at this year's AstiCon in AZ? Are the tutorials going to be worthwhile for fellow Asterisk users/admins that have been actively building, running, and administering Asterisk boxes for years? I have a suggestion for one demo that would certainly interest me and other more advanced users. I would *love* to see a hands on live demo of this recently posted clustering idea of Vicente Aguilar. http://www.bisente.com/blog/2007/08/26/asterisk-cluster-fonebridge2/?lan=english I would also like to see a tutorial on TDMoE (if that is even supported anymore). I am sure I can think of some more but it is still too early. An outline of what is planned for the tutorial portion of the show would be much appreciated. Thanks, Steve Totaro ___ --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] Prepaid Billing: A2Billing, AstBill, ASTCC
bilal ghayyad wrote: Hi List; I need to use an prepaid billing system with Asterisk, and I do not know which one is more stable and integrated with Asterisk? A2Billing or AstBill or ASTCC? Also, from where I can download it and ready about its configuration? Regards ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 009659849460 Have you looked at ASTPP? Have not looked in a while but Darren had plans to integrate it into OSCommerce and some other neat features. I think he based it on the original ASTCC but has made some major improvements. Just another thing to look at... Thanks, Steve ___ --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] Prepaid Billing: A2Billing, AstBill, ASTCC
On Mon, Aug 27, 2007 at 07:12:02AM -0400, Steve Totaro wrote: Have you looked at ASTPP? Have not looked in a while but Darren had plans to integrate it into OSCommerce and some other neat features. I think he based it on the original ASTCC but has made some major improvements. Does it work with '-T' and 'use strict'? -- 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] No LongDistance for 1 Extension?
This is what I did in Trixbox: I added this to extensions_custom.conf - [restrict-local-only] include = from-internal-additional-custom include = app-recordings include = app-callwaiting-cwoff include = app-callwaiting-cwon include = app-dialvm include = app-vmmain include = app-cf-busy-off include = app-cf-busy-off-any include = app-cf-busy-on include = app-cf-off include = app-cf-off-any include = app-cf-on include = app-cf-unavailable-off include = app-cf-unavailable-on include = ext-meetme include = app-calltrace include = app-directory include = app-echo-test include = app-speakextennum include = app-speakingclock include = app-dnd-off include = app-dnd-on include = app-pickup include = app-chanspy include = ext-test include = ext-local include = outrt-007-local-only include = restrict-invalid exten = h,1,Hangup [restrict-invalid] exten = _9.,1,Playback(feature-not-avail-line) exten = _9.,n,Playback(that-number) exten = _9.,n,Playback(is) exten = _9.,n,Playback(privacy-not) exten = _9.,n,Playback(accessible-through-system) exten = _9.,n,Busy() -- Then in trixbox, each extension has a context field. Change it from the default to restrict-local-only. Also, I added a Route called local-only This includes just our local exchanges, emergency, and toll free. Dial Patterns: 911 9|1248. 9|1576. 9|1713. 9|1800. 9|1810. 9|1866. 9|1877. 9|1888. I created the restrict-invalid context to play a recording when a call was blocked. It matches anything not specified in restrict-local-only or higher included contexts. This scenario work great for me. Supposedly there is a Trixbox module called CustomContexts http://aussievoip.com.au/wiki/freePBX-CustomContexts , but it is in beta and seems more complicated than my approach. It should be much more versatile, but I went with the quick fix. -- -- Steven http://www.glimasoutheast.org Thomas Kenyon [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Seysan wrote: Hi all, I want to limit the outgoing trunk to certain extensions, so for example 6 extensions can call long distance, but 4 other extensions are not allowed to do so. How can I do it in FreePBX specially? I don't know about Trixbox per say, but normally you would have all the handsets that can make long distance calls in one context and all the ones that can't in another, then use dialplan logic to glue it all together. ___ --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] Problem with Page command
What is happening ? Please email us the SIP Debug. Also with paging most phones require a special SIP header for the phone to know that it has to pick up right away. - Original Message - From: Stuart J. Newman To: asterisk-users@lists.digium.com Sent: Monday, August 13, 2007 6:53 PM Subject: [asterisk-users] Problem with Page command I am using the page command per the example in the Wiki and am having trouble getting it to work the way I want. The call is coming from a SipXchange system and all the phones are attached to the SipXchange. Please let me know what config file you need. I also have the sip debug trace available. Stuart J. Newman System Engineer IT Globalsat Telecommunications A Globecomm Systems Company Voice (240) 553-9423 Fax (301) 483-4350 [EMAIL PROTECTED] www.globalsat.com -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --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] Stereo Conferences?
Are there any speakerphones or other conferencing HW phones that play the audio in stereo? Either their own speakers, or jacks for an amp with room speakers? Is there any way for Asterisk to deliver call legs with stereo channels in the RTP stream? If not, is it possible for Asterisk to keep 2 separate calls, or pairs of legs in a conference call, synced exactly enough (including traveling over the Net between the same 2 IP#s) for them to arrive as a stereo pair at the endpoint? -- (C) Matthew Rubenstein ___ --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] Stereo Conferences?
although not stereo i believe its the closest you will get if the codec is supported by asterisk. polycom has now HD codec On 8/27/07, Matthew Rubenstein [EMAIL PROTECTED] wrote: Are there any speakerphones or other conferencing HW phones that play the audio in stereo? Either their own speakers, or jacks for an amp with room speakers? Is there any way for Asterisk to deliver call legs with stereo channels in the RTP stream? If not, is it possible for Asterisk to keep 2 separate calls, or pairs of legs in a conference call, synced exactly enough (including traveling over the Net between the same 2 IP#s) for them to arrive as a stereo pair at the endpoint? -- (C) Matthew Rubenstein ___ --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] Stereo Conferences?
Do any softphones run the HD codec? What exactly is the HD codec technically called, and is there any info about its codec running inside Asterisk? On Mon, 2007-08-27 at 08:47 -0400, C F wrote: although not stereo i believe its the closest you will get if the codec is supported by asterisk. polycom has now HD codec On 8/27/07, Matthew Rubenstein [EMAIL PROTECTED] wrote: Are there any speakerphones or other conferencing HW phones that play the audio in stereo? Either their own speakers, or jacks for an amp with room speakers? Is there any way for Asterisk to deliver call legs with stereo channels in the RTP stream? If not, is it possible for Asterisk to keep 2 separate calls, or pairs of legs in a conference call, synced exactly enough (including traveling over the Net between the same 2 IP#s) for them to arrive as a stereo pair at the endpoint? -- (C) Matthew Rubenstein ___ --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 -- (C) Matthew Rubenstein ___ --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] voip provider settings problem, please help
hi ppl, i'm using asterisk 1.2 because i'm making use of voiceone, but before i was using asterisk 1.4 and had the same problem, it concerns an italian voip/sip provider called eutelia/skypho, my problem is the following one: when i start my pbx my skypho account is working fine, meaning that e.g. incoming calls are shown in the asterisk CLI and caller and callee can hear each other when picked up, but after a while it stops working, incoming calls for this provider are not shown anymore in the CLI, but from other providers it always works, but the phone is ringingn nevertheless when calling my skypho account...when i then turn off the pbx and restart after sumthing like 2 hours my skypho account is working fine again, the incmiong calls are shown in the asterisk CLI, but after, i don't know let's say an hour or so it again stops working, incoming calls for my skypho account can not be seen in the asterisk CLI, then if i turn off the pbx for an hour or so it works again, so i thought it must be a setting issue, maybe something with the register? althought it always shows it registered when i use 'sip show registry' someone has an idea what i have to set or do to have it working permanently? what could be the problem here? i got no clue whatsoever and i have been using asterisk only since half a year, please help me, i'm totaly desperate, thx in advance jody :) Get news delivered with the All new Yahoo! Mail. Enjoy RSS feeds right on your Mail page. Start today at http://mrd.mail.yahoo.com/try_beta?.intl=ca ___ --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] Stereo Conferences?
AFAIK the HD codec they use is the ITU-T G722.2 AKA GSM-AMR-WB, the big improvement here is the sampling rate ( 16kHz ). On 8/27/07, Matthew Rubenstein [EMAIL PROTECTED] wrote: Do any softphones run the HD codec? What exactly is the HD codec technically called, and is there any info about its codec running inside Asterisk? On Mon, 2007-08-27 at 08:47 -0400, C F wrote: although not stereo i believe its the closest you will get if the codec is supported by asterisk. polycom has now HD codec On 8/27/07, Matthew Rubenstein [EMAIL PROTECTED] wrote: Are there any speakerphones or other conferencing HW phones that play the audio in stereo? Either their own speakers, or jacks for an amp with room speakers? Is there any way for Asterisk to deliver call legs with stereo channels in the RTP stream? If not, is it possible for Asterisk to keep 2 separate calls, or pairs of legs in a conference call, synced exactly enough (including traveling over the Net between the same 2 IP#s) for them to arrive as a stereo pair at the endpoint? -- (C) Matthew Rubenstein ___ --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 -- (C) Matthew Rubenstein ___ --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 -- Guilherme Loch Góes MSN:[EMAIL PROTECTED] (48) 99115299 ___ --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] Stereo Conferences?
The codec is G722 I believe. and Polycom has a conference speaker phone with a subwoofer option that has HD voice. On 8/27/07, Matthew Rubenstein [EMAIL PROTECTED] wrote: Do any softphones run the HD codec? What exactly is the HD codec technically called, and is there any info about its codec running inside Asterisk? On Mon, 2007-08-27 at 08:47 -0400, C F wrote: although not stereo i believe its the closest you will get if the codec is supported by asterisk. polycom has now HD codec On 8/27/07, Matthew Rubenstein [EMAIL PROTECTED] wrote: Are there any speakerphones or other conferencing HW phones that play the audio in stereo? Either their own speakers, or jacks for an amp with room speakers? Is there any way for Asterisk to deliver call legs with stereo channels in the RTP stream? If not, is it possible for Asterisk to keep 2 separate calls, or pairs of legs in a conference call, synced exactly enough (including traveling over the Net between the same 2 IP#s) for them to arrive as a stereo pair at the endpoint? -- (C) Matthew Rubenstein ___ --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 -- (C) Matthew Rubenstein ___ --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 -- Bruce Reeves Nortex Networks ___ --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] Stereo Conferences?
The HD Codec is just G.722 /b On Aug 27, 2007, at 7:52 AM, Matthew Rubenstein wrote: Do any softphones run the HD codec? What exactly is the HD codec technically called, and is there any info about its codec running inside Asterisk? On Mon, 2007-08-27 at 08:47 -0400, C F wrote: although not stereo i believe its the closest you will get if the codec is supported by asterisk. polycom has now HD codec On 8/27/07, Matthew Rubenstein [EMAIL PROTECTED] wrote: Are there any speakerphones or other conferencing HW phones that play the audio in stereo? Either their own speakers, or jacks for an amp with room speakers? Is there any way for Asterisk to deliver call legs with stereo channels in the RTP stream? If not, is it possible for Asterisk to keep 2 separate calls, or pairs of legs in a conference call, synced exactly enough (including traveling over the Net between the same 2 IP#s) for them to arrive as a stereo pair at the endpoint? -- (C) Matthew Rubenstein ___ --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 -- (C) Matthew Rubenstein ___ --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] Stereo Conferences?
The 601 has g722 (and its not g722.1 or .2) /b On Aug 27, 2007, at 8:14 AM, Bruce Reeves wrote: The codec is G722 I believe. and Polycom has a conference speaker phone with a subwoofer option that has HD voice. On 8/27/07, Matthew Rubenstein [EMAIL PROTECTED] wrote: Do any softphones run the HD codec? What exactly is the HD codec technically called, and is there any info about its codec running inside Asterisk? On Mon, 2007-08-27 at 08:47 -0400, C F wrote: although not stereo i believe its the closest you will get if the codec is supported by asterisk. polycom has now HD codec On 8/27/07, Matthew Rubenstein [EMAIL PROTECTED] wrote: Are there any speakerphones or other conferencing HW phones that play the audio in stereo? Either their own speakers, or jacks for an amp with room speakers? Is there any way for Asterisk to deliver call legs with stereo channels in the RTP stream? If not, is it possible for Asterisk to keep 2 separate calls, or pairs of legs in a conference call, synced exactly enough (including traveling over the Net between the same 2 IP#s) for them to arrive as a stereo pair at the endpoint? -- (C) Matthew Rubenstein ___ --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 -- (C) Matthew Rubenstein ___ --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 -- Bruce Reeves Nortex Networks ___ --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] Stereo Conferences?
FreeSWITCH supports 16k wideband conferences and supports G.722, speex 16k and should work great with the phones that support it. I have personally tested it with grandstream phones. /b On Aug 27, 2007, at 7:47 AM, C F wrote: although not stereo i believe its the closest you will get if the codec is supported by asterisk. polycom has now HD codec On 8/27/07, Matthew Rubenstein [EMAIL PROTECTED] wrote: Are there any speakerphones or other conferencing HW phones that play the audio in stereo? Either their own speakers, or jacks for an amp with room speakers? Is there any way for Asterisk to deliver call legs with stereo channels in the RTP stream? If not, is it possible for Asterisk to keep 2 separate calls, or pairs of legs in a conference call, synced exactly enough (including traveling over the Net between the same 2 IP#s) for them to arrive as a stereo pair at the endpoint? -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] which OS would be fine for asterisk
On Thu, 2007-03-15 at 18:24 -0400, Steve Totaro wrote: I am bringing up several Fedora Core 7 boxen into production now. Besides a knee jerk reaction that Fedora Sucks, can someone give a real argument as to why I should or should not use it for production? (besides the several MB of yum updates daily, which to me is a good thing). First of all, let me state for the record that I'm a big fan of (and contributor to) Fedora as a desktop Linux distribution. Also, I'm taking my Digium hat off for a minute... these opinions are mine, and not to be confused with any sort of official position from Digium. The biggest problem I see with Fedora (it's no longer called Fedora Core as of version 7 -- it's just Fedora again) as a distro for a PBX is that packages are only updated for at most 13 months. So, for example, many people using Fedora Core 3 for their PBX no longer have access to security updates, etc. for their Asterisk box. They basically assume you're OK with upgrading your box every year, or that you don't care about long-term updates (which may be fine for a desktop machine, but is less friendly in terms of a server OS). Personally, I use CentOS (when I don't care about support) or RHEL (when support is important to me) as my preferred server distribution, simply because they guarantee to have *years* worth (at least five years!) of security updates, even if I choose not to upgrade to the latest distribution. (Debian has a similar policy, although I'm not sure the exact length of time.) As an added bonus, most of the server-class hardware vendors (HP, Dell, IBM, etc.) seem to have better driver support for RHEL than any other distribution. They might have a slower release cycle (averaging 18 to 24 months) than Fedora (which is averaging 6-7 months between releases), but the long-term viability makes the trade-off worth it in my mind. In the end though, it really boils down to this: The best Linux distribution for your Asterisk box is the one you are the most comfortable, especially when it comes to making sure the box is stable and secure. -Jared Smith ___ --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] voip provider settings problem, please help
Am Montag, den 27.08.2007, 08:55 -0400 schrieb Jody Gugelhupf: hi ppl, i'm using asterisk 1.2 because i'm making use of voiceone, but before i was using asterisk 1.4 and had the same problem, it concerns an italian voip/sip provider called eutelia/skypho, my problem is the following one: when i start my pbx my skypho account is working fine, meaning that e.g. incoming calls are shown in the asterisk CLI and caller and callee can hear each other when picked up, but after a while it stops working, incoming calls for this provider are not shown anymore in the CLI, but from other providers it always works, but the phone is ringingn nevertheless when calling my skypho account...when i then turn off the pbx and restart after sumthing like 2 hours my skypho account is working fine again, the incmiong calls are shown in the asterisk CLI, but after, i don't know let's say an hour or so it again stops working, incoming calls for my skypho account can not be seen in the asterisk CLI, then if i turn off the pbx for an hour or so it works again, so i thought it must be a setting issue, maybe something with the register? althought it always shows it registered when i use 'sip show registry' someone has an idea what i have to set or do to have it working permanently? what could be the problem here? i got no clue whatsoever and i have been using asterisk only since half a year, please help me, i'm totaly desperate, thx in advance jody :) Jody, you could post the relevant parts of your sip.conf here. For me (with a similar problem) introducing qualify=yes to the provider context in sip.conf solved the problem about 99.9% of the time; about three times a week I am off for less than 5 minutes at one particular providers - others work fine (I have a cronjob checking asterisk -rx sip show registry | grep 022396whatever which reports if status is NOT Registered - it does not do anything if the peer is not registered except sending me a notifier mail, so I have some kind of tracking). I am not familiar with italian voiceone though. Best, Anselm ___ --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] error in linking libmfcr2
hi, I am using debian 4.0 with version 2.6.18-4-686 I have downloaded the required files form site asterisk-1.2.24.tar.gz libmfcr2-0.0.3-1.4.tar.bz2 libsupertone-0.0.2.tar.gz libunicall-0.0.3-1.4.tar.bz2 spandsp-20060903.tar.gz I downloaded and installed the files in the follwing sequence spandsp libsupertone libunicall Till here it is compiling and copying .so library to /usr/local/lib/ libmfcr2-0.0.3 source code by doing some modifications compiled correctly but while linking it is giving error libtool: link: only absolute run ptahs are allowed. when running make command Can anybody tell me how to overcome this error Thanks and regards sanchal ___ --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't create audio conversation between softphonesthrough Asterisk
Hi, In the early stages of deciding how to try and develop this environment, I looked at all the protocols that could be used. SIP was chosen just because it seemed to me that it was the most widely used protocol. I believe IAX is a new protocol with a little less documentation and examples. The good thing about this Jain-sip-phone is that it saves a lot of time since many of the important classes are more or less written already. In short, my goal is to create a custom softphone GUI interface. I am using this Jain-sip-phone as an example, so that I could learn the SIP protocol/RTP transmission better. I have not really started altering much of the code yet because I was trying to see if it would run as is, so I have not tried dialing the Jain clients without a subscription. I believe Asterisk does accept subscription requests, but for some reason it doesn't like this one. I will soon start to experiment with the source code. Thanks, Denis -Original Message- From: Gerald A [mailto:[EMAIL PROTECTED] Sent: Monday, August 27, 2007 9:30 AM To: Kutman [EMAIL PROTECTED](Mat) DAEPM(RCS)@Ottawa-Hull Subject: Re: [asterisk-users] Can't create audio conversation between softphonesthrough Asterisk Hi, On 8/27/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Thanks for the reply. I have a small LAN network which I have connected with an Asterisk server. My Asterisk box and the user pc's are connected through a LAN switch. This network is not connected to the internet. The UNREACHABLE message does seem to point to what you mentioned below (Asterisk not being able to ping the phones), which seems weird to me. When I use X-Lite softphones on those user pc's, I can connect them to the Asterisk server fine and make calls. The subscription occurs when I try to add another contact(In the same LAN network) from one of the user pc's. I am attaching the console results that I get within Eclipse when I run this softphone. Ok, one more silly question -- might it be possible to do this with IAX? (I tend to lean on IAX for things, as it's more versitile and robust, if not so widely deployed). I'm not sure exactly what you are trying to accomplish, so I'm focusing on the questions you are having issues with. A bit of context might show up as another solution, though -- if you are able to provide it. I don't have time right now to dig through the traces, but I have a related question. Have you ever got a call to go through dialling from one Jain client to the other, without the subscription? My gut feeling is that there might be a basic config issue with the Jain client that is causing an issue, as what you want to do doesn't sound too difficult. Thanks, Gerald. ___ --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] Tuning a ZyWALL for Asterisk
Anyone? On Aug 24, 2007, at 3:35 PM, Ed Pastore wrote: I understand this question is over-broad, but hopefully you can have patience with a newbie and toss me a bone... I am in the testing stage of deploying Asterisk. I have successfully configured it to work behind the NAT of my ZyXEL ZyWALL 35 firewall. However, I think there is a lot of tuning I can do to get better reliability, bandwidth management, and maybe QoS from the firewall. I have some clues as to how to do some of this, but both telephony and routing are not strong points for me (I mostly work on systems, servers, and LANs). Is there any sort of reference material that will guide me in setting up my ZyWALL for VoIP? I don't see much help from ZyXEL, and I only see scattered posts around the net, but I know a lot of people are using ZyWALLs with Asterisk. If there isn't a reference, then can anyone chime in with some particulars on what you've done? Any hints would be greatly appreciated. Thanks! ___ --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
[asterisk-users] console/dsp 1.4.11
I have an entry for console/dsp in the dialplan. When I call into that extension I get connected to the soundcard and I hear myself etc... everything is fine. However, if I call in and get connected then a second call comes in they also get connected. I was expecting them to get a busy signal or something... Do I have something configured wrong? How do I only get 1 person at a time on console/dsp? Thanks, Jerry ___ --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] console/dsp 1.4.11
Jerry Geis wrote:
However, if I call in and get connected then a second call comes in
they also get connected.
I was expecting them to get a busy signal or something...
Your dialplan needs to take this into account. I do the following:
; *
; Check database entry to see if paging is active, if YES skip to line 6
; else continue on to line 3. We don't want 2 or more active pages
; *
exten = s,1,Set(active=${DB(paging/active)})
exten = s,2,GotoIf($[${active} = YES]?6:3)
; ***
; Set database entry for paging active to YES
; ***
exten = s,3,Set(DB(paging/active)=YES)
;
; Start recording to paging.gsm, no longer then 30 seconds
; If silence for 5 seconds, terminate recording
;
exten = s,4,Record(paging:gsm|5|30)
exten = s,5,Hangup()
;
; If paging currently in use, jump to paging-inuse
; context.
;
exten = s,6,Goto(paging-inuse,s,1)
;
; On hangup from paging, run the pagemerge script
; then set paging/active to NO.
;
exten = h,1,System(/usr/local/bin/pagemerge.sh)
exten = h,2,Set(DB(paging/active)=NO)
[paging-inuse]
exten = s,1,Congestion
exten = s,2,Hangup()
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
___
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Re: [asterisk-users] Calling Clients or Tele Marketing
On 8/26/07, Seysan [EMAIL PROTECTED] wrote: Hello, Let's say I have a Database of my clients about 50 clients, I want to announce a new product or service to them, can asterisk do it for me? It is something like a appointment reminder for doctors. I want to know is there any software for this or I should Write a program for it using AGI or ruby on Rails. You can do a mass-dialing by using .call files, or sending manager actions. All you need is a script that writes one (or several at time) call file once per 10 minutes, going trough list of all your customer numbers. And in asterisk dialplan just play recorded message. For more information see: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out Regards, Atis -- Atis Lezdins, IT Responsible of BEST Riga, [EMAIL PROTECTED] ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? - www.BEST.eu.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] AstriCon Tutorials
Steve Totaro wrote: Can someone outline what tutorials will be covered at this year's AstiCon in AZ? Here is what is available so far: http://www.astricon.net/files/2007-astricon-schedule.pdf -- 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] No LongDistance for 1 Extension?
Thank you. Now that the conexts are different can all the extension call to echother ? Seysan On 8/27/07, Steven [EMAIL PROTECTED] wrote: This is what I did in Trixbox: I added this to extensions_custom.conf - [restrict-local-only] include = from-internal-additional-custom include = app-recordings include = app-callwaiting-cwoff include = app-callwaiting-cwon include = app-dialvm include = app-vmmain include = app-cf-busy-off include = app-cf-busy-off-any include = app-cf-busy-on include = app-cf-off include = app-cf-off-any include = app-cf-on include = app-cf-unavailable-off include = app-cf-unavailable-on include = ext-meetme include = app-calltrace include = app-directory include = app-echo-test include = app-speakextennum include = app-speakingclock include = app-dnd-off include = app-dnd-on include = app-pickup include = app-chanspy include = ext-test include = ext-local include = outrt-007-local-only include = restrict-invalid exten = h,1,Hangup [restrict-invalid] exten = _9.,1,Playback(feature-not-avail-line) exten = _9.,n,Playback(that-number) exten = _9.,n,Playback(is) exten = _9.,n,Playback(privacy-not) exten = _9.,n,Playback(accessible-through-system) exten = _9.,n,Busy() -- Then in trixbox, each extension has a context field. Change it from the default to restrict-local-only. Also, I added a Route called local-only This includes just our local exchanges, emergency, and toll free. Dial Patterns: 911 9|1248. 9|1576. 9|1713. 9|1800. 9|1810. 9|1866. 9|1877. 9|1888. I created the restrict-invalid context to play a recording when a call was blocked. It matches anything not specified in restrict-local-only or higher included contexts. This scenario work great for me. Supposedly there is a Trixbox module called CustomContexts http://aussievoip.com.au/wiki/freePBX-CustomContexts , but it is in beta and seems more complicated than my approach. It should be much more versatile, but I went with the quick fix. -- -- Steven http://www.glimasoutheast.org Thomas Kenyon [EMAIL PROTECTED] wrote in message news: [EMAIL PROTECTED] Seysan wrote: Hi all, I want to limit the outgoing trunk to certain extensions, so for example 6 extensions can call long distance, but 4 other extensions are not allowed to do so. How can I do it in FreePBX specially? I don't know about Trixbox per say, but normally you would have all the handsets that can make long distance calls in one context and all the ones that can't in another, then use dialplan logic to glue it all together. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling Clients or Tele Marketing
Hi, Then, If the number we called was busy, or he didn't pick up the phone, we should call him again. how we can keep track of those ? On 8/27/07, Atis [EMAIL PROTECTED] wrote: On 8/26/07, Seysan [EMAIL PROTECTED] wrote: Hello, Let's say I have a Database of my clients about 50 clients, I want to announce a new product or service to them, can asterisk do it for me? It is something like a appointment reminder for doctors. I want to know is there any software for this or I should Write a program for it using AGI or ruby on Rails. You can do a mass-dialing by using .call files, or sending manager actions. All you need is a script that writes one (or several at time) call file once per 10 minutes, going trough list of all your customer numbers. And in asterisk dialplan just play recorded message. For more information see: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out Regards, Atis -- Atis Lezdins, IT Responsible of BEST Riga, [EMAIL PROTECTED] ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? - www.BEST.eu.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 ___ --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] which OS would be fine for asterisk
On Thu, 2007-03-15 at 18:24 -0400, Steve Totaro wrote: [snip] Besides a knee jerk reaction that Fedora Sucks, can someone give a real argument as to why I should or should not use it for production? (besides the several MB of yum updates daily, which to me is a good thing). Steve, Fedora 7 supports High Resolution Timers which (afaik) is not present in the RHEL5/CentOS5 kernels. If I understand it correctly this could be beneficial on a box that has no TDM card. Guess you could test the difference and see if it is beneficial for your setup. The patch for ztdummy which improved zttest results for me can be found here: http://bugs.digium.com/view.php?id=10314 Regards, Patrick ___ --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] Calling Clients or Tele Marketing
On 8/27/07, Seysan [EMAIL PROTECTED] wrote: Hi, Then, If the number we called was busy, or he didn't pick up the phone, we should call him again. how we can keep track of those ? It's all described in link i gave. There are MaxRetries and RetryTime parameters available, and you can also use 'failed' extension. Regards, Atis -- Atis Lezdins, IT Responsible of BEST Riga, [EMAIL PROTECTED] ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? - www.BEST.eu.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] Calling Clients or Tele Marketing
On 8/27/07, Atis [EMAIL PROTECTED] wrote: On 8/27/07, Seysan [EMAIL PROTECTED] wrote: Hi, Then, If the number we called was busy, or he didn't pick up the phone, we should call him again. how we can keep track of those ? It's all described in link i gave. There are MaxRetries and RetryTime parameters available, and you can also use 'failed' extension. Just while scrolling that page, i noticed a ready dialplan that might do what you need. http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out+deliver+message Regards, Atis -- Atis Lezdins, IT Responsible of BEST Riga, [EMAIL PROTECTED] ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? - www.BEST.eu.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] Calling Clients or Tele Marketing
thank you On 8/27/07, Atis [EMAIL PROTECTED] wrote: On 8/27/07, Atis [EMAIL PROTECTED] wrote: On 8/27/07, Seysan [EMAIL PROTECTED] wrote: Hi, Then, If the number we called was busy, or he didn't pick up the phone, we should call him again. how we can keep track of those ? It's all described in link i gave. There are MaxRetries and RetryTime parameters available, and you can also use 'failed' extension. Just while scrolling that page, i noticed a ready dialplan that might do what you need. http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out+deliver+message Regards, Atis -- Atis Lezdins, IT Responsible of BEST Riga, [EMAIL PROTECTED] ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? - www.BEST.eu.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 ___ --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] which OS would be fine for asterisk
On 8/27/07, Jared Smith [EMAIL PROTECTED] wrote: Personally, I use CentOS (when I don't care about support) or RHEL (when support is important to me) as my preferred server distribution, simply because they guarantee to have *years* worth (at least five years!) of security updates, even if I choose not to upgrade to the latest distribution. (Debian has a similar policy, although I'm not sure the exact length of time.) Debian usually provides regular updates until next major release release, and security updates within year after next major release. Plus a really good thing is that major releases come out with interval of 2 til 5 years - so they are much better tested than all the other distributions (with release cycle of half year). Also upgrade to next version is usually painless (i have seen some troubles with Debian's fork project - Ubuntu). So, if you are into long-term stability and regular updates - Debian have it. However for desktop i prefer Gentoo. It also have very good policy about updates - you don't have to worry much about them when you find right tools. But i don't want my servers to be busy with regular compiling - so servers are Debian. Regards, Atis -- Atis Lezdins, IT Responsible of BEST Riga, [EMAIL PROTECTED] ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? - www.BEST.eu.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] Can't create audio conversation between softphonesthrough Asterisk
Hi, On 8/27/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: In the early stages of deciding how to try and develop this environment, I looked at all the protocols that could be used. SIP was chosen just because it seemed to me that it was the most widely used protocol. I believe IAX is a new protocol with a little less documentation and examples. The good thing about this Jain-sip-phone is that it saves a lot of time since many of the important classes are more or less written already. In short, my goal is to create a custom softphone GUI interface. I am using this Jain-sip-phone as an example, so that I could learn the SIP protocol/RTP transmission better. The reason I asked is because IAX works better through firewalls and is easier to troubleshoot. It's not as widely deployed as SIP, but it does work around some major things that SIP makes harder. I'm not sure of the quality or lineage of the JAIN application code, so can't comment if it's a good jumping off point. I have not really started altering much of the code yet because I was trying to see if it would run as is, so I have not tried dialing the Jain clients without a subscription. I believe Asterisk does accept subscription requests, but for some reason it doesn't like this one. I will soon start to experiment with the source code. Subscription is used for presence. It can be used in an IM type app, or to light up a button on a phone when someone is busy. It shouldn't be needed to exchange a call though, and if you can do it without the subscription piece then it could help to pin down the issue you are having. (It might be _just_ the subscribe that is having an issue). I should have time later this afternoon to check your traces, and I'll try and give Jain a kick. Thanks, Gerald. ___ --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] PRI cards, Digium vs. Sangoma
They know what they are doing and do a lot of it. I don't have to give an opinion myself. There is enough evidence all over for people to draw the proper conclusions for themselves. -Original Message- From: C F [mailto:[EMAIL PROTECTED] Sent: Sunday, August 26, 2007 4:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI cards, Digium vs. Sangoma On 8/26/07, shadowym [EMAIL PROTECTED] wrote: Well there are a couple fine examples of FUD if I do say so myself. Just do a search and see what cards the 'serious' companies out there are using. Nuff said. Can you define 'serious'? -Original Message- From: Doug Lytle [mailto:[EMAIL PROTECTED] Sent: Sunday, August 26, 2007 8:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI cards, Digium vs. Sangoma Eric ManxPower Wieling wrote: Sangoma cards are complicated to set up, have a history of kernel (and zaptel) VERSION issues. i.e. It seems like the zaptel or kernel version I'm running on a machine is always something newer than is supported by the Sangoma drivers. Never had any issues once I got it compiled. You're forgetting one. I'm terrified of upgrading zaptel or the kernel from remote with the systems I have Sangoma cards on. I have, on many occasions, had kernel panics when trying to shut down wanrouter. I don't have this 'fear' with Digium cards. 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] PRI cards, Digium vs. Sangoma
And the FUD continues. Pain and misery eh? Google pain misery insertmodel#here? -Original Message- From: Steve Underwood [mailto:[EMAIL PROTECTED] Sent: Sunday, August 26, 2007 5:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI cards, Digium vs. Sangoma shadowym wrote: Well there are a couple fine examples of FUD if I do say so myself. Just do a search and see what cards the 'serious' companies out there are using. Nuff said. When did saying nothing at all become enough? Eric ManxPower Wieling wrote: Sangoma cards are complicated to set up, have a history of kernel (and zaptel) VERSION issues. i.e. It seems like the zaptel or kernel version I'm running on a machine is always something newer than is supported by the Sangoma drivers. Never had any issues once I got it compiled. Yep. Right now they can't work with the 2.6.22 kernel. I know annoying kernel changes cause them trouble, but they don't respond with the speed they should. Most people want to keep their platforms fully updated, and for many that means the 2.6.22 kernel is going onto their system around now. They also keep poor notes. For example, when 3.1.0 became necessary to be able to use a recent kernel, finding out that is was necessary took some effort. You're forgetting one. I'm terrified of upgrading zaptel or the kernel from remote with the systems I have Sangoma cards on. I have, on many occasions, had kernel panics when trying to shut down wanrouter. I don't have this 'fear' with Digium cards. I've never had a panic, but I have no expectation of a smooth ride when updating the Sangoma drivers. Pain and misery is a more the norm. It is best to wipe out everything you can find on your machine related to wanpipe and zaptel before an upgrade. They seems to end up using bits of old material under some circumstances, causing strange results. Regards, Steve ___ --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't create audio conversation betweensoftphonesthrough Asterisk
Thanks very much for the help, I appreciate it. Recently, one of my co-workers and I have altered the code to just register with the Asterisk server and place an audio call. This gets rid of the subscription part of the application, so I do not get the 489 Bad Event error anymore. I believe the 488 Not Acceptable Here error occurs when the invite is being sent. After the sdp body and header information are created, they are sent as an invite for the audio call. The problem seems to be some part of the invite that we are sending. I have a hunch that it may have to do with the codecs that the Jain-phone chooses. I will continue looking into this. Denis -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Gerald A Sent: Monday, August 27, 2007 2:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't create audio conversation betweensoftphonesthrough Asterisk Hi, On 8/27/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: In the early stages of deciding how to try and develop this environment, I looked at all the protocols that could be used. SIP was chosen just because it seemed to me that it was the most widely used protocol. I believe IAX is a new protocol with a little less documentation and examples. The good thing about this Jain-sip-phone is that it saves a lot of time since many of the important classes are more or less written already. In short, my goal is to create a custom softphone GUI interface. I am using this Jain-sip-phone as an example, so that I could learn the SIP protocol/RTP transmission better. The reason I asked is because IAX works better through firewalls and is easier to troubleshoot. It's not as widely deployed as SIP, but it does work around some major things that SIP makes harder. I'm not sure of the quality or lineage of the JAIN application code, so can't comment if it's a good jumping off point. I have not really started altering much of the code yet because I was trying to see if it would run as is, so I have not tried dialing the Jain clients without a subscription. I believe Asterisk does accept subscription requests, but for some reason it doesn't like this one. I will soon start to experiment with the source code. Subscription is used for presence. It can be used in an IM type app, or to light up a button on a phone when someone is busy. It shouldn't be needed to exchange a call though, and if you can do it without the subscription piece then it could help to pin down the issue you are having. (It might be _just_ the subscribe that is having an issue). I should have time later this afternoon to check your traces, and I'll try and give Jain a kick. Thanks, Gerald. ___ --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] Heavy duty environment - Is TDM2400P suits?
Steve Totaro wrote: Andrew Joakimsen wrote: On 8/21/07, Chris Mason (Lists) [EMAIL PROTECTED] wrote: Steve Totaro wrote: You should have no problems. Make sure you put surge protection and ground your POTS lines. It is a small investment. I have had SEVERAL FXO modules die or behave strangely after thunderstorms. I cannot prove it was a surge, but logic would indicate that was the issue. Steve, How are you providing surge protection? I have lost a couple of cards to storms also. I did not ground them properly (they declined that option in the original sales process) and they were from installations over one or two years ago. Since it was not really Digium's fault, I did not even bother with the RMA process. I may have tried if they were only a couple of months old. I just bought new modules and billed the customer parts and labor. I also sold them and installed proper grounding and surge supression. Thanks, Steve Totaro ___ --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 Steve, what is the proper grounding procedure and what does good grounding achieve? In this instance we are in an office tower and have 13 POTS lines. There is a good ground hanging off the wall from the legacy system but what do I connect it to? Is a surge/lightning protector product required or is there something more rudimentary can be done? The concern here is more with call quality than lightning/surge protection. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Authenticating SIP user in LDAP database instead of SIP.conf file
Dear Mr Galvin, As of today I am using the res_config_ldap of Astirectory in my test Asterisk system to connect to a test LDAP database of my University. Things seem to be working fine so far. Now I'm faced with the task of installing this in the productive system. Before doing so, I'd sure like to consider trying the RealTime database driver that you people have developed. Why so? because I trust your judgment. I see it is res_config_ldap. You'd be much better using the latest version in the bug tracker. This would mean removing Astirectory module, installing the new driver and loading the new schema into LDAP. In my view, the latter part shouldn't be a concern because the old attributes and object classes (Astirectory) should in no way interfere with the new ones. Besides the old object classes could be deleted from LDAP. Also the former part shouldn't be of much concern either. My only concern as of now is in the installation of the RealTime database driver because the 'readme' file does not say anything about the installation. It only says about the configuration after installation. From the link: http://svn.digium.com/svn/asterisk/team/group/res_config_ldap/ Would it be sufficiant if I were to copy the makefile and res_config_ldap.c to the res/ directory of my running Asterisk and do make; make install? or do I have to do LIBS=-lldap export LIBS ./configure before that? My asterisk version is 1.2.6. Thanks in advance, Abhishek * * * * On 8/27/07, Gavin Henry [EMAIL PROTECTED] wrote: I see it is res_config_ldap. You'd be much better using the latest version in the bug tracker. On 27/08/07, Gavin Henry [EMAIL PROTECTED] wrote: On 26/08/07, Abhishek M S [EMAIL PROTECTED] wrote: Dear Mr Galvin, Gavin ;-) Thank you for the links. Had gone through the bug tracker before though. I was specifically referring to the schema for the driver 'Astirectory' and not the one related to the real time LDAP driver for Open LDAP. It's for any LDAP Compliant Directory Server. In the 'Astirectory' documentation there's a file defining the schema for LDAP which is incomplete. By incomplete I mean the Syntax and few other fields are not defined let alone the schema being a static file. I do understand that for Open LDAP a static file schema should be defined. Not really. in the RealTime driver you can specify which LDAP attributes map to which Asterisk Config settings. The only reason why I preferred Astirectory over the LDAP real time driver was the fact that there is no mapping required for SIP users and peers. OK, maybe I need to go and read more about Astirectory. Regards Abhishek On 8/24/07, Gavin Henry [EMAIL PROTECTED] wrote: Please see the official tracker in the Digium buglist: http://bugs.digium.com/view.php?id=5768 Here are the schemas we did for OpenLDAP: http://bugs.digium.com/file_download.php?file_id=14842type=bug http://bugs.digium.com/file_download.php?file_id=14841type=bug Also, for Novell eDirectory, see: http://forge.voicerd.org/frs/?group_id=7release_id=17 Gavin. -- http://www.suretecsystems.com/services/openldap/ ___ --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 -- http://www.suretecsystems.com/services/openldap/ -- http://www.suretecsystems.com/services/openldap/ ___ --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
[asterisk-users] Detecting tones
Hello folks, I'm interested in detecting tones on specific frequencies with specific timing; for example, I'd like Asterisk to dial out and when the channel starts/call connects, listen for a 1200Hz tone that plays for 100ms. Is this doable with Asterisk using something already extant? After looking through documentation, mailing lists, and some of the source I had the idea that I might be better off using EAGI for this, and coding the actual listener in C. If EAGI were the right way to go, would I be able to respond/send tones back (e.g., DTMF tones) on the audio stream? Or would it go to STDOUT from the EAGI app's perspective? Thanks and cheers, Robert Prince ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No LongDistance for 1 Extension?
Dear all, I'm faced with a similar situation of segregating users in 3 different categories to be able to make: internal calls only (students); internal local calls (staff); and internal, local international calls (profs). I do understand that 3 different contexts would have to be defined in the extensions.conf file. Are there custom context modules for Asterisk 1.2.6version as well? If not, I'd really appreciate any suggestions or help in this regard. thanks, Abhishek On 8/27/07, Seysan [EMAIL PROTECTED] wrote: Thank you. Now that the conexts are different can all the extension call to echother ? Seysan On 8/27/07, Steven [EMAIL PROTECTED] wrote: This is what I did in Trixbox: I added this to extensions_custom.conf - [restrict-local-only] include = from-internal-additional-custom include = app-recordings include = app-callwaiting-cwoff include = app-callwaiting-cwon include = app-dialvm include = app-vmmain include = app-cf-busy-off include = app-cf-busy-off-any include = app-cf-busy-on include = app-cf-off include = app-cf-off-any include = app-cf-on include = app-cf-unavailable-off include = app-cf-unavailable-on include = ext-meetme include = app-calltrace include = app-directory include = app-echo-test include = app-speakextennum include = app-speakingclock include = app-dnd-off include = app-dnd-on include = app-pickup include = app-chanspy include = ext-test include = ext-local include = outrt-007-local-only include = restrict-invalid exten = h,1,Hangup [restrict-invalid] exten = _9.,1,Playback(feature-not-avail-line) exten = _9.,n,Playback(that-number) exten = _9.,n,Playback(is) exten = _9.,n,Playback(privacy-not) exten = _9.,n,Playback(accessible-through-system) exten = _9.,n,Busy() -- Then in trixbox, each extension has a context field. Change it from the default to restrict-local-only. Also, I added a Route called local-only This includes just our local exchanges, emergency, and toll free. Dial Patterns: 911 9|1248. 9|1576. 9|1713. 9|1800. 9|1810. 9|1866. 9|1877. 9|1888. I created the restrict-invalid context to play a recording when a call was blocked. It matches anything not specified in restrict-local-only or higher included contexts. This scenario work great for me. Supposedly there is a Trixbox module called CustomContexts http://aussievoip.com.au/wiki/freePBX-CustomContexts , but it is in beta and seems more complicated than my approach. It should be much more versatile, but I went with the quick fix. -- -- Steven http://www.glimasoutheast.org Thomas Kenyon [EMAIL PROTECTED] wrote in message news: [EMAIL PROTECTED] Seysan wrote: Hi all, I want to limit the outgoing trunk to certain extensions, so for example 6 extensions can call long distance, but 4 other extensions are not allowed to do so. How can I do it in FreePBX specially? I don't know about Trixbox per say, but normally you would have all the handsets that can make long distance calls in one context and all the ones that can't in another, then use dialplan logic to glue it all together. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Authenticating SIP user in LDAP database instead of SIP.conf file
On 27/08/07, Abhishek M S [EMAIL PROTECTED] wrote: Dear Mr Galvin, Gavin! ;-) As of today I am using the res_config_ldap of Astirectory in my test Asterisk system to connect to a test LDAP database of my University. Things seem to be working fine so far. Now I'm faced with the task of installing this in the productive system. Before doing so, I'd sure like to consider trying the RealTime database driver that you people have developed. Why so? because I trust your judgment. Thanks, but you should still test it yourself. I see it is res_config_ldap. You'd be much better using the latest version in the bug tracker. This would mean removing Astirectory module, installing the new driver and loading the new schema into LDAP. In my view, the latter part shouldn't be a concern because the old attributes and object classes (Astirectory) should in no way interfere with the new ones. Besides the old object classes could be deleted from LDAP. Also the former part shouldn't be of much concern either. Nope, you are correct. My only concern as of now is in the installation of the RealTime database driver because the 'readme' file does not say anything about the installation. It only says about the configuration after installation. From the link: http://svn.digium.com/svn/asterisk/team/group/res_config_ldap/ Would it be sufficiant if I were to copy the makefile and res_config_ldap.c to the res/ directory of my running Asterisk and do make; make install? or do I have to do LIBS=-lldap export LIBS ./configure before that? My asterisk version is 1.2.6. This Digium version is for 1.4.x, not 1.2 Thanks in advance, Abhishek On 8/27/07, Gavin Henry [EMAIL PROTECTED] wrote: I see it is res_config_ldap. You'd be much better using the latest version in the bug tracker. On 27/08/07, Gavin Henry [EMAIL PROTECTED] wrote: On 26/08/07, Abhishek M S [EMAIL PROTECTED] wrote: Dear Mr Galvin, Gavin ;-) Thank you for the links. Had gone through the bug tracker before though. I was specifically referring to the schema for the driver 'Astirectory' and not the one related to the real time LDAP driver for Open LDAP. It's for any LDAP Compliant Directory Server. In the 'Astirectory' documentation there's a file defining the schema for LDAP which is incomplete. By incomplete I mean the Syntax and few other fields are not defined let alone the schema being a static file. I do understand that for Open LDAP a static file schema should be defined. Not really. in the RealTime driver you can specify which LDAP attributes map to which Asterisk Config settings. The only reason why I preferred Astirectory over the LDAP real time driver was the fact that there is no mapping required for SIP users and peers. OK, maybe I need to go and read more about Astirectory. Regards Abhishek On 8/24/07, Gavin Henry [EMAIL PROTECTED] wrote: Please see the official tracker in the Digium buglist: http://bugs.digium.com/view.php?id=5768 Here are the schemas we did for OpenLDAP: http://bugs.digium.com/file_download.php?file_id=14842type=bug http://bugs.digium.com/file_download.php?file_id=14841type=bug Also, for Novell eDirectory, see: http://forge.voicerd.org/frs/?group_id=7release_id=17 Gavin. -- http://www.suretecsystems.com/services/openldap/ ___ --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 -- http://www.suretecsystems.com/services/openldap/ -- http://www.suretecsystems.com/services/openldap/ ___ --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 -- http://www.suretecsystems.com/services/openldap/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
[asterisk-users] OT: DELL Platforms
Hello list, I have a customer who is interested in standardizing on dell servers for asterisk deployments. Has anyone had success with a particular configuration? Anything specifically to watch out for? Thank you for your time, Art Arthur Miller Sr. Sales Associate VoIP Supply, LLC. 454 Sonwil Drive Buffalo, NY 14225 716-250-3871 OFFICE 716-630-1548 FAX [EMAIL PROTECTED] blocked::mailto:[EMAIL PROTECTED] NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA. ___ --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] OT: DELL Platforms
I have used both the powedge line for large deployments and the Optiplex N series for small offices. The only thing I have had to add to the pc's is 12 power extensions at times and here lately I have had a pc or 2 without the 4 pin molex connector so I had to find SATA to molex adapters. On 8/27/07, Arthur Miller [EMAIL PROTECTED] wrote: Hello list, I have a customer who is interested in standardizing on dell servers for asterisk deployments. Has anyone had success with a particular configuration? Anything specifically to watch out for? Thank you for your time, Art Arthur Miller Sr. Sales Associate VoIP Supply, LLC. 454 Sonwil Drive Buffalo, NY 14225 716-250-3871 OFFICE 716-630-1548 FAX [EMAIL PROTECTED] NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA. ___ --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 -- Bruce Reeves Nortex Networks ___ --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't create audio conversation betweensoftphonesthrough Asterisk
Hi, On 8/27/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Thanks very much for the help, I appreciate it. Recently, one of my co-workers and I have altered the code to just register with the Asterisk server and place an audio call. This gets rid of the subscription part of the application, so I do not get the 489 Bad Event error anymore. I believe the 488 Not Acceptable Here error occurs when the invite is being sent. After the sdp body and header information are created, they are sent as an invite for the audio call. The problem seems to be some part of the invite that we are sending. I have a hunch that it may have to do with the codecs that the Jain-phone chooses. I will continue looking into this. Glad to hear you were able to get some traction with the voice calling. Is the presence bit something that is critical to your custom app? I'm going to be fiddling with some soft phone stuff soon, so I am still planning on taking a peek at Jain just for the heck of it. Keep me updated on your progress, and if you need any assistance, give me a shout. Thanks, Gerald. ___ --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] OT: DELL Platforms
Arthur Miller wrote: Hello list, I have a customer who is interested in standardizing on dell servers for asterisk deployments. Has anyone had success with a particular configuration? Anything specifically to watch out for? Thank you for your time, Art **Arthur Miller** Sr. Sales Associate **VoIP Supply, LLC**. 454 Sonwil Drive Buffalo, NY 14225 716-250-3871 OFFICE 716-630-1548 FAX [EMAIL PROTECTED] blocked::mailto:[EMAIL PROTECTED] I am running an SC1435 with two dual core Opteron 2212, four gigs of RAM and a couple 250gig SATA drives. Totally VoIP so I cannot comment on cards or interrupts, but so far it has been flawless. I would like to see how many G729/ULAW conversions it could handle. How would I go about benchmarking that? Thanks, Steve ___ --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] OT: DELL Platforms
Steve Totaro wrote: Arthur Miller wrote: Hello list, I have a customer who is interested in standardizing on dell servers for asterisk deployments. Has anyone had success with a particular configuration? Anything specifically to watch out for? Thank you for your time, Art **Arthur Miller** Sr. Sales Associate **VoIP Supply, LLC**. 454 Sonwil Drive Buffalo, NY 14225 716-250-3871 OFFICE 716-630-1548 FAX [EMAIL PROTECTED] blocked::mailto:[EMAIL PROTECTED] I am running an SC1435 with two dual core Opteron 2212, four gigs of RAM and a couple 250gig SATA drives. Totally VoIP so I cannot comment on cards or interrupts, but so far it has been flawless. I would like to see how many G729/ULAW conversions it could handle. How would I go about benchmarking that? Thanks, Steve Drooling... processor : 0 vendor_id : AuthenticAMD cpu family : 15 model : 65 model name : Dual-Core AMD Opteron(tm) Processor 2212 HE stepping: 2 cpu MHz : 2000.000 cache size : 1024 KB physical id : 0 siblings: 2 core id : 0 cpu cores : 2 fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt rdtscp lm 3dnowext 3dnow pni cx16 lahf_lm cmp_legacy svm extapic cr8_legacy bogomips: 4002.32 TLB size: 1024 4K pages clflush size: 64 cache_alignment : 64 address sizes : 40 bits physical, 48 bits virtual power management: ts fid vid ttp tm stc processor : 1 vendor_id : AuthenticAMD cpu family : 15 model : 65 model name : Dual-Core AMD Opteron(tm) Processor 2212 HE stepping: 2 cpu MHz : 2000.000 cache size : 1024 KB physical id : 1 siblings: 2 core id : 0 cpu cores : 2 fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt rdtscp lm 3dnowext 3dnow pni cx16 lahf_lm cmp_legacy svm extapic cr8_legacy bogomips: 4002.32 TLB size: 1024 4K pages clflush size: 64 cache_alignment : 64 address sizes : 40 bits physical, 48 bits virtual power management: ts fid vid ttp tm stc processor : 2 vendor_id : AuthenticAMD cpu family : 15 model : 65 model name : Dual-Core AMD Opteron(tm) Processor 2212 HE stepping: 2 cpu MHz : 2000.000 cache size : 1024 KB physical id : 0 siblings: 2 core id : 1 cpu cores : 2 fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt rdtscp lm 3dnowext 3dnow pni cx16 lahf_lm cmp_legacy svm extapic cr8_legacy bogomips: 4002.32 TLB size: 1024 4K pages clflush size: 64 cache_alignment : 64 address sizes : 40 bits physical, 48 bits virtual power management: ts fid vid ttp tm stc processor : 3 vendor_id : AuthenticAMD cpu family : 15 model : 65 model name : Dual-Core AMD Opteron(tm) Processor 2212 HE stepping: 2 cpu MHz : 2000.000 cache size : 1024 KB physical id : 1 siblings: 2 core id : 1 cpu cores : 2 fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt rdtscp lm 3dnowext 3dnow pni cx16 lahf_lm cmp_legacy svm extapic cr8_legacy bogomips: 4002.32 TLB size: 1024 4K pages clflush size: 64 cache_alignment : 64 address sizes : 40 bits physical, 48 bits virtual power management: ts fid vid ttp tm stc ___ --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] Authenticating SIP user in LDAP database instead of SIP.conf file
Dear Mr Gavin, Sorry for having miss pelt your name twice... Thank you once again for your prompt reply. Is this the correct version of the driver for Asterisk 1.2.x : res_config_ldap-v0.7.tar.gzhttp://bugs.digium.com/file_download.php?file_id=9565type=bug from the link http://bugs.digium.com/view.php?id=5768 Thank you for your time and patience, Abhishek On 8/27/07, Gavin Henry [EMAIL PROTECTED] wrote: On 27/08/07, Abhishek M S [EMAIL PROTECTED] wrote: Dear Mr Galvin, Gavin! ;-) As of today I am using the res_config_ldap of Astirectory in my test Asterisk system to connect to a test LDAP database of my University. Things seem to be working fine so far. Now I'm faced with the task of installing this in the productive system. Before doing so, I'd sure like to consider trying the RealTime database driver that you people have developed. Why so? because I trust your judgment. Thanks, but you should still test it yourself. I see it is res_config_ldap. You'd be much better using the latest version in the bug tracker. This would mean removing Astirectory module, installing the new driver and loading the new schema into LDAP. In my view, the latter part shouldn't be a concern because the old attributes and object classes (Astirectory) should in no way interfere with the new ones. Besides the old object classes could be deleted from LDAP. Also the former part shouldn't be of much concern either. Nope, you are correct. My only concern as of now is in the installation of the RealTime database driver because the 'readme' file does not say anything about the installation. It only says about the configuration after installation. From the link: http://svn.digium.com/svn/asterisk/team/group/res_config_ldap/ Would it be sufficiant if I were to copy the makefile and res_config_ldap.c to the res/ directory of my running Asterisk and do make; make install? or do I have to do LIBS=-lldap export LIBS ./configure before that? My asterisk version is 1.2.6. This Digium version is for 1.4.x, not 1.2 Thanks in advance, Abhishek On 8/27/07, Gavin Henry [EMAIL PROTECTED] wrote: I see it is res_config_ldap. You'd be much better using the latest version in the bug tracker. On 27/08/07, Gavin Henry [EMAIL PROTECTED] wrote: On 26/08/07, Abhishek M S [EMAIL PROTECTED] wrote: Dear Mr Galvin, Gavin ;-) Thank you for the links. Had gone through the bug tracker before though. I was specifically referring to the schema for the driver 'Astirectory' and not the one related to the real time LDAP driver for Open LDAP. It's for any LDAP Compliant Directory Server. In the 'Astirectory' documentation there's a file defining the schema for LDAP which is incomplete. By incomplete I mean the Syntax and few other fields are not defined let alone the schema being a static file. I do understand that for Open LDAP a static file schema should be defined. Not really. in the RealTime driver you can specify which LDAP attributes map to which Asterisk Config settings. The only reason why I preferred Astirectory over the LDAP real time driver was the fact that there is no mapping required for SIP users and peers. OK, maybe I need to go and read more about Astirectory. Regards Abhishek On 8/24/07, Gavin Henry [EMAIL PROTECTED] wrote: Please see the official tracker in the Digium buglist: http://bugs.digium.com/view.php?id=5768 Here are the schemas we did for OpenLDAP: http://bugs.digium.com/file_download.php?file_id=14842type=bug http://bugs.digium.com/file_download.php?file_id=14841type=bug Also, for Novell eDirectory, see: http://forge.voicerd.org/frs/?group_id=7release_id=17 Gavin. -- http://www.suretecsystems.com/services/openldap/ ___ --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 -- http://www.suretecsystems.com/services/openldap/ -- http://www.suretecsystems.com/services/openldap/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] OT: DELL Platforms
Hi, About 2 years ago we made the decision to ship exclusively Dell servers. Mostly we have shipped the 860 rackmount with a config of a basic dual core proc couple gig of RAM and a pair of 75GB HDDs in RAID 1. And they are great but we put a limit of about 30 concurrent calls through it. That being said we have got larger installs too, we are running 2 of the older 2950's as a fully redundant load balancing pair. For a call center of around 160. The only thing I would watch for is with the 860 the TE110p doesn't work. The TE120p is fantastic no problems but the older card had some incompatibility. Other than that I've never had one skip a beat, so I hope you have the same luck. Cheers, Joel Hill Support Manager Asterisk IT On Mon, 2007-08-27 at 18:15 -0400, Steve Totaro wrote: Steve Totaro wrote: Arthur Miller wrote: Hello list, I have a customer who is interested in standardizing on dell servers for asterisk deployments. Has anyone had success with a particular configuration? Anything specifically to watch out for? Thank you for your time, Art **Arthur Miller** Sr. Sales Associate **VoIP Supply, LLC**. 454 Sonwil Drive Buffalo, NY 14225 716-250-3871 OFFICE 716-630-1548 FAX [EMAIL PROTECTED] blocked::mailto:[EMAIL PROTECTED] I am running an SC1435 with two dual core Opteron 2212, four gigs of RAM and a couple 250gig SATA drives. Totally VoIP so I cannot comment on cards or interrupts, but so far it has been flawless. I would like to see how many G729/ULAW conversions it could handle. How would I go about benchmarking that? Thanks, Steve Drooling... processor : 0 vendor_id : AuthenticAMD cpu family : 15 model : 65 model name : Dual-Core AMD Opteron(tm) Processor 2212 HE stepping: 2 cpu MHz : 2000.000 cache size : 1024 KB physical id : 0 siblings: 2 core id : 0 cpu cores : 2 fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt rdtscp lm 3dnowext 3dnow pni cx16 lahf_lm cmp_legacy svm extapic cr8_legacy bogomips: 4002.32 TLB size: 1024 4K pages clflush size: 64 cache_alignment : 64 address sizes : 40 bits physical, 48 bits virtual power management: ts fid vid ttp tm stc processor : 1 vendor_id : AuthenticAMD cpu family : 15 model : 65 model name : Dual-Core AMD Opteron(tm) Processor 2212 HE stepping: 2 cpu MHz : 2000.000 cache size : 1024 KB physical id : 1 siblings: 2 core id : 0 cpu cores : 2 fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt rdtscp lm 3dnowext 3dnow pni cx16 lahf_lm cmp_legacy svm extapic cr8_legacy bogomips: 4002.32 TLB size: 1024 4K pages clflush size: 64 cache_alignment : 64 address sizes : 40 bits physical, 48 bits virtual power management: ts fid vid ttp tm stc processor : 2 vendor_id : AuthenticAMD cpu family : 15 model : 65 model name : Dual-Core AMD Opteron(tm) Processor 2212 HE stepping: 2 cpu MHz : 2000.000 cache size : 1024 KB physical id : 0 siblings: 2 core id : 1 cpu cores : 2 fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt rdtscp lm 3dnowext 3dnow pni cx16 lahf_lm cmp_legacy svm extapic cr8_legacy bogomips: 4002.32 TLB size: 1024 4K pages clflush size: 64 cache_alignment : 64 address sizes : 40 bits physical, 48 bits virtual power management: ts fid vid ttp tm stc processor : 3 vendor_id : AuthenticAMD cpu family : 15 model : 65 model name : Dual-Core AMD Opteron(tm) Processor 2212 HE stepping: 2 cpu MHz : 2000.000 cache size : 1024 KB physical id : 1 siblings: 2 core id : 1 cpu cores : 2 fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt rdtscp lm 3dnowext 3dnow pni cx16 lahf_lm cmp_legacy svm extapic cr8_legacy bogomips: 4002.32 TLB size: 1024 4K pages clflush size: 64 cache_alignment : 64 address
Re: [asterisk-users] AsteriskNOW Web GUI
Will this work on 1.2.x? I just installed it and did make samples. The README references a file called html.conf which does not exist and also abruptly ends with the word to on a blank line. Besides that, what would the URL be for AsteriskNow? Is that customizable in the elusive html.conf file? Any GUIs that are easily installed on existing systems and work with 1.2.x? Thanks, Steve bkruse wrote: svn co http://svn.digium.com/svn/asterisk-gui/branches/asterisknow thegui; cd thegui; sh configure; make sudo make install ; clear ; echo 'completed' -bk Yann JOUANIN wrote: You can do it from svn server , I think there is a page in the wiki Best, yann *De :* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *De la part de* Jeremy Mann *Envoyé :* vendredi 24 août 2007 17:30 *À :* Asterisk Users Mailing List - Non-Commercial Discussion *Objet :* [asterisk-users] AsteriskNOW Web GUI Is the web GUI for AsteriskNOW able to be loaded on an existing server(that was installed from ubuntu-server and asterisk loaded from source)? ___ --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] AsteriskNOW Web GUI
On Mon, Aug 27, 2007 at 07:38:37PM -0400, Steve Totaro wrote: Will this work on 1.2.x? I just installed it and did make samples. Yeah. Just backport support for the manager over http, users.conf, and a few other small things. (read: no). -- 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] OT: DELL Platforms
I've run up to 50 concurrent calls on the PE850 and PE860 using TE205p. I also came across the te110p issue which manifests itself as popping and crackling audio. It is rather insidious as zttest is fine, the problem does not appear to be missed interrupts. In my case the Digium distributor refused to take back the card (we were within the 30 day return period), so I only buy Sangoma now. Craig -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joel Hill Sent: Tuesday, 28 August 2007 7:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: DELL Platforms Hi, About 2 years ago we made the decision to ship exclusively Dell servers. Mostly we have shipped the 860 rackmount with a config of a basic dual core proc couple gig of RAM and a pair of 75GB HDDs in RAID 1. And they are great but we put a limit of about 30 concurrent calls through it. That being said we have got larger installs too, we are running 2 of the older 2950's as a fully redundant load balancing pair. For a call center of around 160. The only thing I would watch for is with the 860 the TE110p doesn't work. The TE120p is fantastic no problems but the older card had some incompatibility. Other than that I've never had one skip a beat, so I hope you have the same luck. Cheers, Joel Hill Support Manager Asterisk IT On Mon, 2007-08-27 at 18:15 -0400, Steve Totaro wrote: Steve Totaro wrote: Arthur Miller wrote: Hello list, I have a customer who is interested in standardizing on dell servers for asterisk deployments. Has anyone had success with a particular configuration? Anything specifically to watch out for? Thank you for your time, Art **Arthur Miller** Sr. Sales Associate **VoIP Supply, LLC**. 454 Sonwil Drive Buffalo, NY 14225 716-250-3871 OFFICE 716-630-1548 FAX [EMAIL PROTECTED] blocked::mailto:[EMAIL PROTECTED] I am running an SC1435 with two dual core Opteron 2212, four gigs of RAM and a couple 250gig SATA drives. Totally VoIP so I cannot comment on cards or interrupts, but so far it has been flawless. I would like to see how many G729/ULAW conversions it could handle. How would I go about benchmarking that? Thanks, Steve Drooling... processor : 0 vendor_id : AuthenticAMD cpu family : 15 model : 65 model name : Dual-Core AMD Opteron(tm) Processor 2212 HE stepping: 2 cpu MHz : 2000.000 cache size : 1024 KB physical id : 0 siblings: 2 core id : 0 cpu cores : 2 fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt rdtscp lm 3dnowext 3dnow pni cx16 lahf_lm cmp_legacy svm extapic cr8_legacy bogomips: 4002.32 TLB size: 1024 4K pages clflush size: 64 cache_alignment : 64 address sizes : 40 bits physical, 48 bits virtual power management: ts fid vid ttp tm stc processor : 1 vendor_id : AuthenticAMD cpu family : 15 model : 65 model name : Dual-Core AMD Opteron(tm) Processor 2212 HE stepping: 2 cpu MHz : 2000.000 cache size : 1024 KB physical id : 1 siblings: 2 core id : 0 cpu cores : 2 fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt rdtscp lm 3dnowext 3dnow pni cx16 lahf_lm cmp_legacy svm extapic cr8_legacy bogomips: 4002.32 TLB size: 1024 4K pages clflush size: 64 cache_alignment : 64 address sizes : 40 bits physical, 48 bits virtual power management: ts fid vid ttp tm stc processor : 2 vendor_id : AuthenticAMD cpu family : 15 model : 65 model name : Dual-Core AMD Opteron(tm) Processor 2212 HE stepping: 2 cpu MHz : 2000.000 cache size : 1024 KB physical id : 0 siblings: 2 core id : 1 cpu cores : 2 fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt rdtscp lm 3dnowext 3dnow pni cx16 lahf_lm cmp_legacy svm extapic cr8_legacy bogomips: 4002.32 TLB size: 1024 4K pages clflush size: 64 cache_alignment : 64 address sizes : 40 bits physical, 48 bits virtual power management: ts fid vid ttp tm stc processor : 3 vendor_id : AuthenticAMD cpu family : 15 model : 65
[asterisk-users] app-conference
Hi: I think app-conference is used where there isn't zaptel hardware,in the other word when we use zaptel hardware we shouldn't use app-conference for conference call sevice and we should use meetme application and load ztdummy.Is it true? Best regards. - Pinpoint customers who are looking for what you sell. ___ --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] app-conference
Hi: I think app-conference is used where there isn't zaptel hardware,in the other word when we use zaptel hardware we shouldn't use app-conference for conference call sevice and we should use meetme application and load ztdummy.Is it true? Best regards. - Shape Yahoo! in your own image. Join our Network Research Panel today!___ --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
