[asterisk-users] e164 Format Numbers
This is probably a very simple question, but I can't for the life of me work it out. I'm trying to use Asterisk as a PTSN gateway to OCS (and believe I have all the SIP issues sorted), but OCS wants to dial in e164 format (+613blahblah). Because Asterisk sees the "+" in the SIP URI, it doesn't want to match anything in my dial plan, not even the S extension in the nominated context. Am I missing something completely obvious? ___ -- 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] Best phone for easy provisioning
Does anyone have any recommendations for a phone that has easy to understand/implement central provisioning? I've used CISCO 79XX phones, and they're great (but too expensive). I like Grandstream phones, but their provisioning sucks. What is everybody else using in large environments where individual config is not an option? ---- Rod Bacon Technical Manager JASCO Consulting Pty. Ltd. http://www.jasco.net.au <http://www.jasco.net.au/> Ph. 03 9432 6376 Fax: 03 9432 6378 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Realtime UPDATE
I'm sorry if this has been answered before, but I've been through all the (lengthy) threads on Realtime, and can't find the answer. My problem is that upon registration, the UA's IP address and port information isn't being written to the MYSQL realtime database. Subsequently, calls to the UA fail if they originate from another * server (The server DOES attempt a lookup, but obviously gets no value for IP address / PORT). From the MYSQL logs, I see the folowing at registration; UPDATE sip SET ipaddr = '', port = '', regseconds = '0', username = '9998' WHERE name = '9998' The weird thing is that it was working at some point yesterday. Can anyone suggest a place to start looking? Also, how do I enable debug logging so I can see the realtime info in the * CLI or logs? -- == Rod Bacon - VOIP Systems Engineer Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime UPDATE
I found the logging stuff. Here is the * debug info. Apr 7 09:09:20 DEBUG[3672]: res_config_mysql.c:117 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip WHERE name = '' Apr 7 09:09:20 DEBUG[3672]: res_config_mysql.c:605 mysql_reconnect: MySQL RealTime: Everything is fine. -- SIP Seeding peers from Astdb: '' at [EMAIL PROTECTED]:5060 for 3600 Apr 7 09:09:20 DEBUG[3672]: res_config_mysql.c:117 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip WHERE name = '' Apr 7 09:09:20 DEBUG[3672]: res_config_mysql.c:605 mysql_reconnect: MySQL RealTime: Everything is fine. -- SIP Seeding peers from Astdb: '' at [EMAIL PROTECTED]:5060 for 3600 -- Unregistered SIP '' Apr 7 09:09:20 DEBUG[3672]: res_config_mysql.c:311 update_mysql: MySQL RealTime: Update SQL: UPDATE sip SET ipaddr = '', port = '', regseconds = '0', username = '' WHERE name = '' Apr 7 09:09:20 DEBUG[3672]: res_config_mysql.c:605 mysql_reconnect: MySQL RealTime: Everything is fine. Apr 7 09:09:20 DEBUG[3672]: res_config_mysql.c:330 update_mysql: MySQL RealTime: Updated 0 rows on table: sip Apr 7 09:09:20 DEBUG[3672]: res_config_mysql.c:117 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip WHERE name = '' Apr 7 09:09:20 DEBUG[3672]: res_config_mysql.c:605 mysql_reconnect: MySQL RealTime: Everything is fine. Apr 7 09:09:20 DEBUG[3672]: db.c:177 ast_db_get: Unable to find key '' in family 'SIP/Registry' Apr 7 09:09:20 DEBUG[3672]: res_config_mysql.c:117 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip WHERE name = '' Apr 7 09:09:20 DEBUG[3672]: res_config_mysql.c:605 mysql_reconnect: MySQL RealTime: Everything is fine. Apr 7 09:09:20 DEBUG[3672]: db.c:177 ast_db_get: Unable to find key '' in family 'SIP/Registry' -- Registered SIP '' at 192.168.0.137 port 5060 expires 3600 -- Saved useragent "Grandstream BT100 1.0.5.22" for peer Apr 7 09:09:20 DEBUG[3672]: res_config_mysql.c:311 update_mysql: MySQL RealTime: Update SQL: UPDATE sip SET ipaddr = '', port = '', regseconds = '0', username = '' WHERE name = '' Apr 7 09:09:20 DEBUG[3672]: res_config_mysql.c:605 mysql_reconnect: MySQL RealTime: Everything is fine. Apr 7 09:09:20 DEBUG[3672]: res_config_mysql.c:330 update_mysql: MySQL RealTime: Updated 0 rows on table: sip Apr 7 09:09:20 DEBUG[3672]: res_config_mysql.c:117 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip WHERE name = '' Apr 7 09:09:20 DEBUG[3672]: res_config_mysql.c:605 mysql_reconnect: MySQL RealTime: Everything is fine. -- SIP Seeding peers from Astdb: '' at [EMAIL PROTECTED]:5060 for 3600 Apr 7 09:09:35 DEBUG[3672]: chan_sip.c:937 __sip_autodestruct: Auto destroying call '[EMAIL PROTECTED]' - Original Message - From: "Rod Bacon" <[EMAIL PROTECTED]> To: Sent: Thursday, April 07, 2005 9:06 AM Subject: [Asterisk-Users] Realtime UPDATE I'm sorry if this has been answered before, but I've been through all the (lengthy) threads on Realtime, and can't find the answer. My problem is that upon registration, the UA's IP address and port information isn't being written to the MYSQL realtime database. Subsequently, calls to the UA fail if they originate from another * server (The server DOES attempt a lookup, but obviously gets no value for IP address / PORT). From the MYSQL logs, I see the folowing at registration; UPDATE sip SET ipaddr = '', port = '', regseconds = '0', username = '9998' WHERE name = '9998' The weird thing is that it was working at some point yesterday. Can anyone suggest a place to start looking? Also, how do I enable debug logging so I can see the realtime info in the * CLI or logs? -- == Rod Bacon - VOIP Systems Engineer Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime UPDATE
I had discovered this myself. Once I set this value, the updates started to occur, but as shown in my earlier post, are all NULL values. - Original Message - From: "Thierry Wehr" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Thursday, April 07, 2005 9:36 AM Subject: RE: [Asterisk-Users] Realtime UPDATE -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Rod Bacon Envoyé : jeudi 7 avril 2005 01:06 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] Realtime UPDATE My problem is that upon registration, the UA's IP address and port information isn't being written to the MYSQL realtime database. Subsequently, calls to the UA fail if they originate from another * server (The server DOES attempt a lookup, but obviously gets no value for IP address / PORT). Hi I'd the same problem and discovered that you have to put rtnoupdate=no in you'r sip.conf Hope it helps you Thierry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to avoid that certain calls come into thevoicemail (e.g. wakeup calls)?
It also looks to me, on the test I just ran, that it will try calling twice, even though MaxRetries is set to 1. That's right. If you only want it to call once, set max REtries to 0. -Andy On Apr 6, 2005 5:30 PM, Ronald Wiplinger <[EMAIL PROTECTED]> wrote: We use wakeup calls for reminders, but it happens, that the person to be reminded is on the phone. To get a voicemail later is not really useful anymore, ... Is there a way to avoid that? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No more updates of IP address and port in CVS HEAD
Bugger... That explains it... - Original Message - From: "Thierry Wehr" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Sunday, March 20, 2005 2:08 AM Subject: [Asterisk-Users] No more updates of IP address and port in CVS HEAD Good afternoon Since the cvs version of yesterday, the ip address and the port of the sipfriend are no more updated in the realtime database Regards Thierry Wehr ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP - SIP Problems
I'd personally be using Ethereal to look inside the SIP messages for the SDP info and checking the source/destination of the resultant RTP stream. One-way audio is typical of NAT issues. Although you are running a VPN (of sorts) I suspect that your SDP messages are getting screwed up somewhere. What are the asterisk NAT settings in effect for each of the SIP phones? I'd be inclined to turn them both ON to ensure that symmetrical RTP in being used. Also make sure that canreinvite is OFF for both. - Original Message - From: "Ian Pattison" <[EMAIL PROTECTED]> To: Sent: Thursday, April 07, 2005 4:49 AM Subject: [Asterisk-Users] SIP - SIP Problems Hi Everybody... Continuing the litany of problems I'm experiencing with my new system I'm =etting issues connecting between SIP phones. A bit of background... I have an asterisk server running in a central =ocation where I have two incoming analog lines connected to FXO ports, =wo analog phones connecting to FXS ports and a single SIP phone. In =ddition I have a remote site connected via a CIPE VPN (ok..ok I know it's =ot a real VPN...) with a single SIP phone. Calls initiated from the remote SIP phone to the central SIP phone often =ave trouble... the user of the central phone cannot hear anything from =he remote phone although everything is heard at the remote phone. If the =emote phone calls either outside or to one of the Zap phones there is no =rouble. If the local SIP phone calls the remote SIP phone there is no =rouble. Both phones are from the same vendor, have the same firmware and =he same configuration with the exception of phone number, PIN, IP address =tc. What am I doing wrong here? Ian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Softphone for testing with Asterisk
I've tested about a dozen of them, and find firefly one of the best (others have more features, but I find firefly is a good mix of quality/features/performance). Make sure you get the third-party firefly though, not the one that's limited to virbiage. Try here... http://www.virbiage.com/firefly/download/firefly-thirdparty.exe - Original Message - From: "raymond" <[EMAIL PROTECTED]> To: Sent: Thursday, April 07, 2005 1:41 PM Subject: [Asterisk-Users] SIP Softphone for testing with Asterisk Hi all, I had just set up my asterisk server. Can anybody know that is there any sip softphone for testing with asterisk? (I had download some from internet but I think all are preconfig to certain server). Cheers. Raymond ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime UPDATE
I have playing with these, to no effect. I am assuming that threre is indeed a bug. Arnaud PIGNARD wrote: Got the same problem, and i rollback to older version (before RT cache patch). There is a combinaison where you will get successfull update but for me, realtime was unstable. I haven't yet time to make more test for maybe report a potential bug. Try modify theses params to yes or no : Rtcachefriends=yes Rtnoupdate=yes Rtautoclear=yes At 01:39 07/04/2005, you wrote: I had discovered this myself. Once I set this value, the updates started to occur, but as shown in my earlier post, are all NULL values. - Original Message - From: "Thierry Wehr" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Thursday, April 07, 2005 9:36 AM Subject: RE: [Asterisk-Users] Realtime UPDATE -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Rod Bacon Envoyé : jeudi 7 avril 2005 01:06 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] Realtime UPDATE My problem is that upon registration, the UA's IP address and port information isn't being written to the MYSQL realtime database. Subsequently, calls to the UA fail if they originate from another * server (The server DOES attempt a lookup, but obviously gets no value for IP address / PORT). Hi I'd the same problem and discovered that you have to put rtnoupdate=no in you'r sip.conf Hope it helps you Thierry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- == Rod Bacon - VOIP Systems Engineer Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Beeps during Sip to Sip phone calls
Me either... - Original Message - From: "Cameron Beattie" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, April 07, 2005 4:07 PM Subject: Re: [Asterisk-Users] Beeps during Sip to Sip phone calls I have a SPA2000 and haven't noticed this. However I haven't used it extensively. Regards Cameron - Original Message - From: "Me" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, April 07, 2005 5:17 PM Subject: Re: [Asterisk-Users] Beeps during Sip to Sip phone calls Anyone else using Sipura equipment and having excessive BEEPing? Maybe a firmware upgrade would help? -- Wholesale Private Label Internet Access! http://www.YourOwnISP.com - Original Message - From: "Rich Adamson" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, April 07, 2005 12:52 AM Subject: Re: [Asterisk-Users] Beeps during Sip to Sip phone calls Inline... I keep hearing DTMF type beeps when on phone calls, I know this is some sort of trait of VOIP but it's driving me nuts.. Not really. I noticed that it happens MUCH more when I am on the phone with one particular person. We are using SPA-2000's from Sipura on both ends. I'm using a spa-3000 and have noticed the same thing. Some voices trigger it, others don't. It hasn't happened often enough to cause me to spend time on it. (I have the spa3000 configured so that incoming fxo calls go directly to the fxs port (not through *), so in my case the dtmf-like bursts have to be internal spa issues. Since the spa2000 and 3000 share a lot of the same code base, the tones you're hearing are likely internal spa issues as well.) Tonight I was looking at the CLI (*command line interface) while I was on the phone with this person. Each time I heard a beep, I saw at EXACTLY the same time the following line: -- Attempting native bridge of SIP/206-5286 and SIP/109-fbf7 What's wierd is that we were already on the phone, I was 206 and he was 109. Does this give anyone a clue as to what might be happening here? I also saw a bunch of these but not sure if it was related to our call or not. Apr 6 21:16:03 NOTICE[3587]: rtp.c:430 ast_rtp_read: RTP: Received packet with bad UDP checksum Apr 6 21:16:05 NOTICE[3587]: rtp.c:430 ast_rtp_read: RTP: Received packet with bad UDP checksum Apr 6 21:16:39 NOTICE[3587]: rtp.c:430 ast_rtp_read: RTP: Received packet with bad UDP checksum Apr 6 21:16:41 NOTICE[3587]: rtp.c:430 ast_rtp_read: RTP: Received packet with bad UDP checksum Apr 6 21:16:44 NOTICE[3587]: rtp.c:430 ast_rtp_read: RTP: Received packet with bad UDP checksum Apr 6 21:17:03 NOTICE[3587]: rtp.c:430 ast_rtp_read: RTP: Received packet with bad UDP checksum Apr 6 21:17:10 NOTICE[3587]: rtp.c:430 ast_rtp_read: RTP: Received packet with bad UDP checksum I'd guess the above messages are simply damaged packets (eg, ethernet collisions, broadband hits). Since there are multiple seconds between most of those messages, I would doubt that you would actually notice the hits in the audio. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime UPDATE
| ++--+-+--+---+--+-+-+---+--+--++-+--+--+-+---+-++--+--+-+--+-+-+++++--+--+-+-+++--++ - Original Message - From: "Matthew Boehm" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, April 07, 2005 11:31 PM Subject: Re: [Asterisk-Users] Realtime UPDATE Apr 7 09:09:20 DEBUG[3672]: db.c:177 ast_db_get: Unable to find key '' in family 'SIP/Registry' This is the problem. Can you post your 'DESCRIBE ' and the row containing user ? You using any NAT? You NAT column should be varchar(5) not int. -Matthew Rod Bacon wrote: I have playing with these, to no effect. I am assuming that threre is indeed a bug. Arnaud PIGNARD wrote: Got the same problem, and i rollback to older version (before RT cache patch). There is a combinaison where you will get successfull update but for me, realtime was unstable. I haven't yet time to make more test for maybe report a potential bug. Try modify theses params to yes or no : Rtcachefriends=yes Rtnoupdate=yes Rtautoclear=yes At 01:39 07/04/2005, you wrote: I had discovered this myself. Once I set this value, the updates started to occur, but as shown in my earlier post, are all NULL values. - Original Message - From: "Thierry Wehr" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Thursday, April 07, 2005 9:36 AM Subject: RE: [Asterisk-Users] Realtime UPDATE -----Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Rod Bacon Envoyé : jeudi 7 avril 2005 01:06 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] Realtime UPDATE My problem is that upon registration, the UA's IP address and port information isn't being written to the MYSQL realtime database. Subsequently, calls to the UA fail if they originate from another * server (The server DOES attempt a lookup, but obviously gets no value for IP address / PORT). Hi I'd the same problem and discovered that you have to put rtnoupdate=no in you'r sip.conf Hope it helps you Thierry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] APPRADIUS cdr_radius.so
I want to use the appradius module (specifically the cdr_radius.so) module to dump asterisk CDR to a RADIUS server (in addition to the local SQL database). I don't want the authorisation component, only the CDR->RADIUS function. I have downloaded, compiled and installed the software without error, but can't locate a scrap of documentation that indicates how I use the cdr_radius.so module. Has anyone done this? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Off Topic - Employment Opportunity - PERL, Melbourne, AU.
Sorry if this is off-topic, but I know there's a quite a few smart people who frequent these groups, and I was thinking that it'd be a good place to ask. We have an opening for an experienced PERL programmer. If you (or anyone you know) is interested, please feel free to email me for more details. -- ====== Rod Bacon - VOIP Systems Engineer Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Off Topic - Employment Opportunity - PERL, Melbourne, AU.
Sorry, I should have stated that the position is a FULL-TIME position, based in our Melbourne office. - Original Message - From: "Jean-Michel Hiver" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, April 08, 2005 2:55 PM Subject: Re: [Asterisk-Users] Off Topic - Employment Opportunity - PERL,Melbourne, AU. Rod Bacon wrote: Sorry if this is off-topic, but I know there's a quite a few smart people who frequent these groups, and I was thinking that it'd be a good place to ask. We have an opening for an experienced PERL programmer. If you (or anyone you know) is interested, please feel free to email me for more details. Hi, I have 5 years Perl experience, numerous Perl modules on CPAN, and some free time on my hands... - I have written quite a few CPAN modules - I am currently writing AGI scripts I do - Object oriented perl - Like to have proper makefiles, test, and pod that makes things maintainable - use strict; - use warnings; Currently I live in Reunion Island but we could start working immediately on a per-project basis. Regards, Jean-Michel. -- Ykoz Un Max - La VoIP en pré-payé! Essayez gratuitement - 5 crédits offerts. ---> http://ykoz.net/voip/max <--- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 4 x ISDN2 hardware...?
It's my opinion that whilst asterisk indeed has some fax capability, it's not a business-grade fax platform. If faxes are indeed as important to your business as you suggest, I'd be inclinded to look for alternatives. - Original Message - From: "Marc" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Monday, April 11, 2005 4:29 PM Subject: [Asterisk-Users] 4 x ISDN2 hardware...? Hi, I've done some testing with asterisk and I must say I'm very impressed by all the features. Now I want to create a production environment and am looking into all the available ISDN cards. The cards I've found are: 1. AVM C4 (1300 euro's) 2. Eicon Diva with 4 ISDN2 ports (even more expensive) 3. Junghanss card with 4 ISDN2 ports (600 euro's) Besides the voice part, I would also like to be able to receive and send faxes. Which card is best? If I understand it correctly, the junghanss card is a 4 port HFC card. I tested Asterisk with another 1 port HFC card and rxfax, but found out that not all faxes are received correctly. As my business is depending on faxes, I find it very important that the incoming faxes are received correctly. Does anybody have experience with receiving faxes with the AVM C4? I assume that it is fully supported by Asterisk and the analog modem to receive faxes as implemented in the hardware of the AVM C4? Any information would be appreciated. Thanks! Marc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Callback application
I don't know if what you're trying to do is possible, but the easiest way to check would be to take a look at the raw packets on the ethernet interface of your * server once a call is in progress. If indeed the RTP can be handed off to the 2 endpoints, you should only see SIP traffic at your server. TCPDUMP is your friend. - Original Message - From: "snacktime" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, April 11, 2005 1:32 PM Subject: [Asterisk-Users] Callback application I wasn't sure how else to label this thread because I'm not sure on the correct terminology to use when decribing what I'm trying to do... I am using livevoip and have a DID with them also, both using SIP. THe big picture is that I'm making a callback application. Right now I'm testing out a couple of things just using DISA. What I'm trying to do is setup a two legged call using * and DISA, with both legs going to/from livevoip, and set the call up in a way where the voice traffic goes straight between livevoip/livevoip once both legs are established. What I don't know is how to tell if I have succeeded in this. Using the following I get both legs up and * say's it's created a native bridge between the two legs. However a 'sip show channels' still shows both channels in *. How do I tell if the voice data is not going through * anymore? Basically once the legs are joined, with one originating from livevoip and one terminating to livevoip, I want my * box out of the picture as far as the voice data stream goes. [outgoing] exten => _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],30,r) [from-livevoip] exten => 800xxx,1,Ringing exten => 800xxx,2,Wait(1) exten => 800xxx,3,Answer exten => 800xxx,4,DISA(no-password|outgoing) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H.323 General Questions
My experience with VOIP to date has surounded SER and Asterisk, SIP and IAX. It would appear as though I am about to be inducted into the world of H.323 and as such I am interested in hearing from anyone who is using Asterisk extensively in a mixed protocol environment, especially in using H.323 to pickup/put down calls from an upstream provider. I have started some reading, but I am not sure where to devote my attention. Essentially what I'm asking is, should I be looking to use Asterisk to route H.323 calls, or would another product be more appropriate? I am likening this scenario to the Asterisk/SER relationship. Should I be using Asterisk in conjunction with a dedicated H.323 Proxy? Thanks in advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zyxel P2000W Finally (Almost) Working
I found new firmware at: ftp://ftp.us.zyxel.com/P2000W/firmware/P2000W_WJ.00.10_Standard.zip The phone is now finally (almost) useful. Still a cheap piece of crap, with new bugs to replace the old, but at least it sort of works now. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zyxel P2000W Finally (Almost) Working
More on this... While the new features are a welcome change, the most annoying of the original problems remain. If anyone knows why I can make one call, then not make or receive another (until I reboot it), I'd like to know. I think it has something to do with a lack of or a malformed BYE. - Original Message - From: "Rod Bacon" <[EMAIL PROTECTED]> To: Sent: Tuesday, April 12, 2005 2:45 PM Subject: [Asterisk-Users] Zyxel P2000W Finally (Almost) Working I found new firmware at: ftp://ftp.us.zyxel.com/P2000W/firmware/P2000W_WJ.00.10_Standard.zip The phone is now finally (almost) useful. Still a cheap piece of crap, with new bugs to replace the old, but at least it sort of works now. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Realtime Friends
Matthew, I got the updates to start working again by ensuring that rtcachefriends=yes. I don't see why this should make a difference, but it does. My understanding was that this parameter only controlled the seeding of the in-memory friends list from the realtime db for purposes of MWI and KeepAlive. I have, however, one remaining issue that I need to resolve. Essentially, I am testing two Asterisk servers (Server1 ans Server2), configured to talk to a common database. I am trying to have calls placed on ANY server routed to SIP UAs registered on ANY OTHER server. Specifically; UA1 registers to Server1. DB is updated correctly. UA2 registers to Server2. DB is updated correctly. I can query the db (using REALTIME LOAD) from either server and see the correct SIP info for either UA. The central dialplan simply routes calls to SIP/UA1 or SIP/UA2. The problem is that Server1 ONLY knows about UA1 and Server2, UA2. The logic seems to be that the lookup in the extensions table (realtime dialplan) happens, then tries to route the call to a SIP registrant that is not in the local (in-memory) friends table. I thought the Server would then go back to the friends realtime table to get the registration info? Is this NOT how it is supposed to work? Should rtcachefriends force the server to update it's friends list on server startup, then at predetermined (configurable?) intervals? Matthew Boehm wrote: (I removed the [] header cause that is what i base my email filter on.) Rod Bacon wrote: I think there's a more sinister bug in play somewhere. The phones are on the same LAN. It was working when I only had a single asterisk server using the database, and seemed to stop when I added a second server. I know this doesn't make any sense... OK. Lemme picture this. You had originally 1 asterisk server and 1 database server. This worked fine with RealTime. Then you added a second asterisk server to connect to this same database server and now the phone won't register with either asterisk server? The SIP registration MUST be ok, because the in-memory database on the server that accepts the registration shows the correct information... the problem is that it doesn't write it to the database. Oh. Weird. But if you turn off the 2nd asterisk server, everything is fine? I think the bug must lie in the update code. When the registration is accepted, the update command is sending nulls to the database for some reason. Yes, this is wierd cause I can't duplicate this. You don't have entries in BOTH sip.conf AND ARA do you? You said the phone does indeed register, it just doesn't update the database using RealTime? Is there any way you can send a full debug output starting slighty before the phone tries to register? have you done a packet sniff to see if asterisk is indeed sending back a 200 OK to the register request? -Matthew -- ====== Rod Bacon - VOIP Systems Engineer Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Local Echo
In addition to making sure that echo cancellation is enabled on the interface(s) in question, you will also need to play with the gain settings. Specifically, try turning down the rxgain. I dropped mine to -10.0, and the echo disappeared altogether. The problem was then that incoming voice was too quiet. After a lot of messing around, I eventually settled on -3.0 This figure gives me good incoming volume and only a faint echo... not enough to bother me or my users. I also found that the order of settings in the zapata.conf makes a difference. If I had the gain settings too far down in the config file, they had no effect. Make sure you stop and restart * after changing any of these settings. A simple reload won't suffice (I even unloaded and reloaded the kernel modules, just to be sure). - Original Message - From: "Jeff Heath" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, April 13, 2005 7:54 AM Subject: Re: [Asterisk-Users] Local Echo Here's what's happening. First some background. Anytime there's a 4 wire (T-1) to 2 wire (local subscriber loop) conversion (this is called a hybrid) there's a good chance that some electrical energy will be reflected. This is because there is usually an impedance mismatch between the 4 wire and 2 wire circuits. This happens all the time in the local telco. You come in to switch A and are destined for switch Z. The telco transports the traffic between A and Z over T-1 (which is muxed up to T-3 or SONET). When the T-1 gets to switch Z it eventually gets attached to a 2 wire local loop (POTS) to get to the far end. Energy from A is reflected back towards A by the hybrid at the Z side. But reflected energy is only one of two necessary conditions for echo. The other condition is sufficient delay for a human being to perceive it as echo. In order for us to perceive it as echo, the reflected energy must be delayed by about 25 msec. Anything less than that and we perceive it as sidetone (sidetone is actually a good thing). The local telephone company doesn't have echo cancelers in their network because they don't need them. Why? because in the local POTS network you'll never have a call that is delayed by more than 25 msec. Long distance carriers (IXCs) install echo cancelers because their customers will experience delays longer than 25 msec, but not local telcos. Now introduce VoIP. VoIP turns every call (even the simple setup you outlined) into a long distance call. If you have your jitter buffer set to 3 you've introduced 60 msec of delay. I forget the rule of thumb for distance vs electrical delay, but I think you can go from NY to SanDiego in about 85 msec. That explains why the echo is there. What I can't help you with (I've got lots of telecom experience, but little Asterisk experience) is changing the settings in Asterisk to cancel it. The good news, though, is that this is a straight-forward echo cancellation problem, and once you find someone who knows what the right settings are, you should be able to get rid of it. -- Jeff Heath On Tue, 2005-04-12 at 17:28, Noah Silverman wrote: Jeff, Thanks for the help. Your explanation of an "echo" makes perfect sense. Here are some notes on our system that might help: 1) The echo occurs on EVERY call either inbound or outbound, local or ld. 2) We don't use any VOIP services, just PTSN lines from the phone company 3) Our system is like this: SIP phone <-> Asterisk box <-> TDM400 card with FXO <-> Telco Pots line 4) I hear my own voice echo. The other party sounds fine to me, and I sound fine to them. 5) The phones are on a very small LAN in our office with almost no traffic. 6) Our phones are Polycom IP500 7) I have the codec set to ulaw Thanks!!! -N Jeff Heath wrote: >On Tue, 2005-04-12 at 15:28, Noah Silverman wrote: > > >>Hi, >> >>I tried, and still get an echo. >>I don't think the problem is with the zap interface. It must be on the >>asterisk or phone side. >> >>-N >> >> >> > >Echo requires 2 phenomena: 1) reflected energy 2) enough delay that it >is discernable. That you are hearing echo means that something at the >far end is reflecting the electrical or accoustical energy of your >voice. > >Echo cancellation should be done as close to the source of unwanted >reflected energy as possible. The fact that you're hearing echo means >that the echo cancelers at the far end either a) don't exist or b) >didn't work. It will be very difficult to cancel reflected energy >coming back at you from the "other side" of the network. > >Tell me more about the phone call where you experienced the echo and I >_might_ be able to help. Specifically, > >- was the phone at the other end a speaker phone and if so, was it an >expensive Polycom phone that's designed to be a speaker phone or a cheap >Walmart phone that happens to have speaker capability? > >- was it a local call or a long distance call > >- what codecs are in use? > >- what's your
Re: [Asterisk-Users] Local Echo
I think that you need to listen to people's advice when you ask for help. I have EXACTLY the same problem as you and I solved it with the methods that I described in my last post. Think about it. How can you get echo on a pure RTP stream from your phone to an asterisk server? Do you hear echo whilst recording voice to your asterisk server (e.g. when leaving a voicemail) or when calling another SIP phone? I think not. Noah Silverman wrote: Hi, I think that you guys are missing the problem. The echo is only from the sidetone. I don't hear the other party with an echo and they don't hear me with an echo. That leads me to believe that it hs nothing to do with the zapata stuff. It is somewhere between my SIP phone as Asterisk. -N Rod Bacon wrote: In addition to making sure that echo cancellation is enabled on the interface(s) in question, you will also need to play with the gain settings. Specifically, try turning down the rxgain. I dropped mine to -10.0, and the echo disappeared altogether. The problem was then that incoming voice was too quiet. After a lot of messing around, I eventually settled on -3.0 This figure gives me good incoming volume and only a faint echo... not enough to bother me or my users. I also found that the order of settings in the zapata.conf makes a difference. If I had the gain settings too far down in the config file, they had no effect. Make sure you stop and restart * after changing any of these settings. A simple reload won't suffice (I even unloaded and reloaded the kernel modules, just to be sure). - Original Message - From: "Jeff Heath" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, April 13, 2005 7:54 AM Subject: Re: [Asterisk-Users] Local Echo Here's what's happening. First some background. Anytime there's a 4 wire (T-1) to 2 wire (local subscriber loop) conversion (this is called a hybrid) there's a good chance that some electrical energy will be reflected. This is because there is usually an impedance mismatch between the 4 wire and 2 wire circuits. This happens all the time in the local telco. You come in to switch A and are destined for switch Z. The telco transports the traffic between A and Z over T-1 (which is muxed up to T-3 or SONET). When the T-1 gets to switch Z it eventually gets attached to a 2 wire local loop (POTS) to get to the far end. Energy from A is reflected back towards A by the hybrid at the Z side. But reflected energy is only one of two necessary conditions for echo. The other condition is sufficient delay for a human being to perceive it as echo. In order for us to perceive it as echo, the reflected energy must be delayed by about 25 msec. Anything less than that and we perceive it as sidetone (sidetone is actually a good thing). The local telephone company doesn't have echo cancelers in their network because they don't need them. Why? because in the local POTS network you'll never have a call that is delayed by more than 25 msec. Long distance carriers (IXCs) install echo cancelers because their customers will experience delays longer than 25 msec, but not local telcos. Now introduce VoIP. VoIP turns every call (even the simple setup you outlined) into a long distance call. If you have your jitter buffer set to 3 you've introduced 60 msec of delay. I forget the rule of thumb for distance vs electrical delay, but I think you can go from NY to SanDiego in about 85 msec. That explains why the echo is there. What I can't help you with (I've got lots of telecom experience, but little Asterisk experience) is changing the settings in Asterisk to cancel it. The good news, though, is that this is a straight-forward echo cancellation problem, and once you find someone who knows what the right settings are, you should be able to get rid of it. -- Jeff Heath On Tue, 2005-04-12 at 17:28, Noah Silverman wrote: Jeff, Thanks for the help. Your explanation of an "echo" makes perfect sense. Here are some notes on our system that might help: 1) The echo occurs on EVERY call either inbound or outbound, local or ld. 2) We don't use any VOIP services, just PTSN lines from the phone company 3) Our system is like this: SIP phone <-> Asterisk box <-> TDM400 card with FXO <-> Telco Pots line 4) I hear my own voice echo. The other party sounds fine to me, and I sound fine to them. 5) The phones are on a very small LAN in our office with almost no traffic. 6) Our phones are Polycom IP500 7) I have the codec set to ulaw Thanks!!! -N Jeff Heath wrote: On Tue, 2005-04-12 at 15:28, Noah Silverman wrote: Hi, I tried, and still get an echo. I don't think the problem is with the zap interface. It must be on the asterisk or phone side. -N Echo requires 2 phenomena: 1) reflected energy 2) enough delay that it is discernable. That you are hearing echo mea
[Asterisk-Users] New SNOM 190 Firmware
The firmware at... http://www.snom.com/download/share/snom190-3.60b-SIP-j.bin seems to have fixed quite a few SNOM 190 bugs. Worth a try. -- == Rod Bacon - VOIP Systems Engineer Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New SNOM 190 Firmware
The one that I immediately noticed was a bug related to "call forward - all" which previously didn't work at all. I also had problems with the web interface pausing for long periods, which also seems to have disappeared. I still have an issue with placing a call on hold and picking it back up. I am testing other features as I type this... - Original Message - From: "Shaun Dwyer" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, April 13, 2005 12:14 PM Subject: Re: [Asterisk-Users] New SNOM 190 Firmware Any ideas what bugs are fixed? there dosn't seem to be any release notes. -Shaun Rod Bacon wrote: The firmware at... http://www.snom.com/download/share/snom190-3.60b-SIP-j.bin seems to have fixed quite a few SNOM 190 bugs. Worth a try. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OH323 and Asterisk CVS-HEAD-03/21/05-15:32:10
I don't know about your * source code, but mine clearly states that you must use OpenH323 v1.15.1 and PWLib v1.8.1 - Original Message - From: "Jose R. Ortiz Ubarri" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, April 14, 2005 2:49 AM Subject: Re: [Asterisk-Users] OH323 and Asterisk CVS-HEAD-03/21/05-15:32:10 I just commented that line and everything is ok. Thanks for help, JO Jose R. Ortiz Ubarri wrote: Yes, I followed the instructions at: http://www.oinko.net/astrecipes/index.php?action=artikel&cat=270174&id=10&artlang=en Guillermo Salas M wrote: On Wed, 2005-04-13 at 08:05, Jose R. Ortiz Ubarri wrote: I have problems compiling the OH323 channel with Asterisk CVS-HEAD-03/21/05-15:32:10. I have the following errors. chan_oh323.c:4895: warning: passing arg 1 of `ast_channel_unregister' from incompatible pointer type chan_oh323.c: In function `load_module': chan_oh323.c:5192: warning: passing arg 1 of `ast_channel_register' from incompatible pointer type chan_oh323.c:5192: error: too many arguments to function `ast_channel_register' make[1]: *** [chan_oh323.o] Error 1 make[1]: Leaving directory `/root/asterisk-oh323-0.7.1/asterisk-driver' make: *** [subdirs_build] Error 1 Have you patched the openh323 code with the file included in asteris-oh323-0.7.1 ? Looks like a compatibility problem with the asterisk functions. Had they changed? I followed the instructions at http://www.oinko.net/astrecipes/index.php?action=artikel&cat=270174&id=10&artlang=en. And I had oh323 working before with a previous version of asterisk... Anyone else had the same problem??? Thanks for help, JO -- Jose R. Ortiz Ubarri (CHEO), CS System Administrator / Programmer High Performance Computing facility - UPR Email: [EMAIL PROTECTED]|[EMAIL PROTECTED] Phone: 787-758-3054 Fax: 787-758-3058 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] does meetme need ztdummy
Yes. It needs a timing source (ztdummy). - Original Message - From: "Xu Wang" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, April 14, 2005 10:34 AM Subject: [Asterisk-Users] does meetme need ztdummy Hello I don't have zaptel installed, but want to run Meetme. I got errors as Apr 13 17:32:53 WARNING[12906]: chan_zap.c:763 zt_open: Unable to open '/dev/zap/pseudo': No such device Apr 13 17:32:53 ERROR[12906]: chan_zap.c:6700 chandup: Unable to dup channel: No such device Apr 13 17:32:53 WARNING[12906]: app_meetme.c:227 build_conf: Unable to open pseudo channel - trying device Apr 13 17:32:54 WARNING[12906]: app_meetme.c:230 build_conf: Unable to open pseudo device Apr 13 17:32:54 DEBUG[12906]: rtp.c:1188 ast_rtp_write: Ooh, format changed from unknown to ulaw -- Playing 'conf-invalid' (language 'en') how to solve it? thank you! steven ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OH323 and Asterisk CVS-HEAD-03/21/05-15:32:10
Sorry for my last post. I see we are talking about different H.323 channel software. - Original Message - From: "Rod Bacon" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, April 14, 2005 8:44 AM Subject: Re: [Asterisk-Users] OH323 and Asterisk CVS-HEAD-03/21/05-15:32:10 I don't know about your * source code, but mine clearly states that you must use OpenH323 v1.15.1 and PWLib v1.8.1 - Original Message - From: "Jose R. Ortiz Ubarri" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, April 14, 2005 2:49 AM Subject: Re: [Asterisk-Users] OH323 and Asterisk CVS-HEAD-03/21/05-15:32:10 I just commented that line and everything is ok. Thanks for help, JO Jose R. Ortiz Ubarri wrote: Yes, I followed the instructions at: http://www.oinko.net/astrecipes/index.php?action=artikel&cat=270174&id=10&artlang=en Guillermo Salas M wrote: On Wed, 2005-04-13 at 08:05, Jose R. Ortiz Ubarri wrote: I have problems compiling the OH323 channel with Asterisk CVS-HEAD-03/21/05-15:32:10. I have the following errors. chan_oh323.c:4895: warning: passing arg 1 of `ast_channel_unregister' from incompatible pointer type chan_oh323.c: In function `load_module': chan_oh323.c:5192: warning: passing arg 1 of `ast_channel_register' from incompatible pointer type chan_oh323.c:5192: error: too many arguments to function `ast_channel_register' make[1]: *** [chan_oh323.o] Error 1 make[1]: Leaving directory `/root/asterisk-oh323-0.7.1/asterisk-driver' make: *** [subdirs_build] Error 1 Have you patched the openh323 code with the file included in asteris-oh323-0.7.1 ? Looks like a compatibility problem with the asterisk functions. Had they changed? I followed the instructions at http://www.oinko.net/astrecipes/index.php?action=artikel&cat=270174&id=10&artlang=en. And I had oh323 working before with a previous version of asterisk... Anyone else had the same problem??? Thanks for help, JO -- Jose R. Ortiz Ubarri (CHEO), CS System Administrator / Programmer High Performance Computing facility - UPR Email: [EMAIL PROTECTED]|[EMAIL PROTECTED] Phone: 787-758-3054 Fax: 787-758-3058 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H.323 in CVS Head
I'm stuck trying to implement H.323 in CVS Head (14 April). The channel driver that comes with Asterisk appears to be broken and another thread talks of the fact that the 3rd party H.323 channel also doesn't work with CVS Head. Still another thread talked of a new H.323 channel being worked on by Digium. Has anyone seen/tested this yet, or is it still vapourware? -- ====== Rod Bacon - VOIP Systems Engineer Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ZyXEL Router Terrible Voice Quality
I have been frustrated by a variety of zyxel issues/products and have found the best solution for all of them lies in a cylindrical receptacle that sits beside my desk... - Original Message - From: "aza" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, April 14, 2005 11:28 AM Subject: [Asterisk-Users] ZyXEL Router Terrible Voice Quality Hi, Has anyone found a solution for the terrible voice quality when using the ZyXEL 2002 units? I have a couple connecting into my asterisk box but the calls are so chopped up and broken that you can't have a conversation. The same router connected to FWD gets crystal clear calls. FWD say they only support G711 which is the codec the calls were using to my asterisk box so it's not a codec issue. Apparently no other settings, apart from the account info were changed, from connecting to FWD to my asterisk box so that would rule out things like silence suppression. Any ideas? Aaron ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel 0 on Zap ???
Idle channels DO restart periodically, so this is fine. You are correct in your assumptions that you need to change your span numbering. I believe it's because your telco is starting at 0 instead of 1. I had this problem also, and had to change mine to span=1,1,0,ccs,hdb3,crc4 Search the list and the wiki... the answer is in there somewhere (that's where I found it). - Original Message - From: Tim Connolly To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Thursday, April 14, 2005 2:14 PM Subject: [Asterisk-Users] Channel 0 on Zap ???
Re: [Asterisk-Users] ztdummy
Conferencing (MeetMe application) - Original Message - From: "Paul Fielding" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, April 14, 2005 12:46 PM Subject: Re: [Asterisk-Users] ztdummy Ok, Here's my ztdummy question. Forgive my ignorance. Everything I read about ztdummy and zaptel cards describes them as being required 'for timing'. But what exactly does this imply? Eg. I have two separate boxes where I did the following: - installed Linux (debian Woody) - compiled a 2.4 kernel - added a few other prereq packages needed to allow Asterisk to compile - compiled and configured Asterisk At this point Asterisk works like a hot darn, no problem, for everything I try to do. No Zaptel card. No ztdummy. So what does ztdummy buy me? regards, Paul - Original Message - From: "[EMAIL PROTECTED]" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, April 13, 2005 5:58 PM Subject: Re: [Asterisk-Users] ztdummy you can get rid of ztdummy. --- Brian Leyton <[EMAIL PROTECTED]> wrote: I installed a couple of Asterisk test machines, and have been successful in getting them talking to one another, but I have question. After installation, I put an x100p clone in one of the machines. From what I understand, I no longer need ztdummy on that machine, but I'm wondering if it hurts anything. If it's better to remove it, where do I go to get rid of it (I'm running [EMAIL PROTECTED] - which uses CentOS, a Redhat variant)? It looks like it's doing something - have a look at the /proc/interrupts: [EMAIL PROTECTED] root]# more /proc/interrupts CPU0 0:1770360 XT-PIC timer 1: 4 XT-PIC keyboard 2: 0 XT-PIC cascade 8: 1 XT-PIC rtc 9: 17667040 XT-PIC wcfxo 11: 17694525 XT-PIC ztdummy, usb-uhci, eth0 12: 19 XT-PIC PS/2 Mouse 14: 13676 XT-PIC ide0 15: 12 XT-PIC ide1 NMI: 0 ERR: 0 Brian Leyton IT Manager Commercial Petroleum Equipment ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RTP not being sent by asterisk
Are the calls coming from SIP or PSTN? - Original Message - From: "trixter http://www.0xdecafbad.com"; <[EMAIL PROTECTED]> To: Sent: Thursday, April 14, 2005 3:56 PM Subject: [Asterisk-Users] RTP not being sent by asterisk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-user ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RTP not being sent by asterisk
Does maximum debugging show anything? - Original Message - From: "trixter http://www.0xdecafbad.com"; <[EMAIL PROTECTED]> To: Sent: Thursday, April 14, 2005 3:56 PM Subject: [Asterisk-Users] RTP not being sent by asterisk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-user ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RTP not being sent by asterisk
From my understanding, * uses the incoming RTP stream itself as a timing source for sending it's outgoing stream, hence the reason * doesn't like/support silence suppression. In other words, if there's no RTP headed back to *, then it won't send anything. (Someone please correct me if I'm talking crap!) I don't know if this is relevant to your situation in any way, but it's worth consideration. trixter http://www.0xdecafbad.com wrote: On Thu, 2005-04-14 at 16:15 +1000, Rod Bacon wrote: Are the calls coming from SIP or PSTN? from sip, and I can see packets going from sip -> asterisk just nothing outside of sip going from asterisk -> sip phone. Its like there is a blocking issue, although I dont know why this would have happened. -- ========== Rod Bacon - VOIP Systems Engineer Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime Friends
Matt, can I assume from your silence that you concurr with my thinking that realtime is in fact broken, or is it that I am using it incorrectly? - Original Message - From: "Rod Bacon" <[EMAIL PROTECTED]> To: "Matthew Boehm" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, April 13, 2005 9:06 AM Subject: [Asterisk-Users] Realtime Friends Matthew, I got the updates to start working again by ensuring that rtcachefriends=yes. I don't see why this should make a difference, but it does. My understanding was that this parameter only controlled the seeding of the in-memory friends list from the realtime db for purposes of MWI and KeepAlive. I have, however, one remaining issue that I need to resolve. Essentially, I am testing two Asterisk servers (Server1 ans Server2), configured to talk to a common database. I am trying to have calls placed on ANY server routed to SIP UAs registered on ANY OTHER server. Specifically; UA1 registers to Server1. DB is updated correctly. UA2 registers to Server2. DB is updated correctly. I can query the db (using REALTIME LOAD) from either server and see the correct SIP info for either UA. The central dialplan simply routes calls to SIP/UA1 or SIP/UA2. The problem is that Server1 ONLY knows about UA1 and Server2, UA2. The logic seems to be that the lookup in the extensions table (realtime dialplan) happens, then tries to route the call to a SIP registrant that is not in the local (in-memory) friends table. I thought the Server would then go back to the friends realtime table to get the registration info? Is this NOT how it is supposed to work? Should rtcachefriends force the server to update it's friends list on server startup, then at predetermined (configurable?) intervals? Matthew Boehm wrote: (I removed the [] header cause that is what i base my email filter on.) Rod Bacon wrote: I think there's a more sinister bug in play somewhere. The phones are on the same LAN. It was working when I only had a single asterisk server using the database, and seemed to stop when I added a second server. I know this doesn't make any sense... OK. Lemme picture this. You had originally 1 asterisk server and 1 database server. This worked fine with RealTime. Then you added a second asterisk server to connect to this same database server and now the phone won't register with either asterisk server? The SIP registration MUST be ok, because the in-memory database on the server that accepts the registration shows the correct information... the problem is that it doesn't write it to the database. Oh. Weird. But if you turn off the 2nd asterisk server, everything is fine? I think the bug must lie in the update code. When the registration is accepted, the update command is sending nulls to the database for some reason. Yes, this is wierd cause I can't duplicate this. You don't have entries in BOTH sip.conf AND ARA do you? You said the phone does indeed register, it just doesn't update the database using RealTime? Is there any way you can send a full debug output starting slighty before the phone tries to register? have you done a packet sniff to see if asterisk is indeed sending back a 200 OK to the register request? -Matthew -- == Rod Bacon - VOIP Systems Engineer Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-user ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dynamic Dialplan - Turn VM on/off?
G'day. I've been working with * for some time now, but mostly from a enterprise perspective. I've just setup my own box at home and want to enable some more "home user" type functionality. Does anyone have a trick to allow the dynamic modification of the dialplan by users? I want the ability to switch voicemail on/off (or at least alter the timeout). In essence, I want to simulate the act of manually turning an answering machine on when you leave home (for my wife). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium G.729 vs. IPP G.729
http://lists.digium.com/pipermail/asterisk-dev/2004-September/006163.html - Original Message - From: Boris Bakchiev To: asterisk-users@lists.digium.com Sent: Monday, April 18, 2005 1:31 PM Subject: [Asterisk-Users] Digium G.729 vs. IPP G.729 Hi, Did anyone compare G.729 implementations (from Digium and the =ne based on IPP) on features, stability, quality and =eliabilty? It would be intresting to know how they fair against each =ther. I could be wrong, but in my testing I did notice a bit more hiss =n Digiums codec thein IPPs. Anyone? Internet communications cannot =e guaranteed to be secured or error-free as information could be =ntercepted, corrupted, lost, destroyed, arrive late or incomplete, or =ontain viruses. Therefore, we do not accept responsibility for any =rrors or omissions that are present in this message, or any attachment, =hat have arisen as a result of e-mail transmission. If verification is =equired, please request a hard-copy version. Any views or opinions =resented are solely those of the author and do not necessarily =epresent those of the company. ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-user ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dynamic callrouting and billing?
I assume you'll be using IAX2 to connect all the servers? In each case, all you need is to match the pattern for the extension then send the call to another * server for final processing. If you only want to maintain this in one place, you could use ARA (Asterisk Realtime Architecture) and store the dialplan in a central database. I've tested this, and it (the dialplan part of ARA) seems to work OK. Given that the call routing will only be 30 lines per server config, I'd probably just manage them in the traditional (distributed, text file based) sense myself. As far as billing goes, we're writing our own system to use the asterisk CDR (stored locally on each server). We haven't determined a roll-up strategy for the databases yet, though being SQL, this is pretty easy to handle. - Original Message - From: "maka" <[EMAIL PROTECTED]> To: Sent: Monday, April 18, 2005 3:35 PM Subject: [Asterisk-Users] dynamic callrouting and billing? Hi everyone, I am trying to figure out a plan for dynamic call forwarding between multiple asterisk servers. I would be dealing with around 30 different extension prefixes, each handled by a distinct asterisk server. Is there a sort of dynamic call routing feature to accomplish this, or I would have to statically describe each extension prefix in extensions.conf (not that it's too much to do any way, but it would be better done dynamically) ? Also, is anyone aware of a free centralized billing solution that I can take a look at so I could possibly start working on my own? Cheers ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hangs pc
This could be any one of about 1.32 million things. Did the PC work OK before you put RH9/Asterisk on it? What sort of BRI card is it? Have you tested the card under another application/OS/platform? What version of Asterisk are you running? Is the BRI card sharing interrupts with anything else? What version of Libpri? What version of Zaptel drivers? What did you eat for dinner last night? What is your favourite sporting team? What is the average velocity of a swallow? A little more information may be helpful. - Original Message - From: "Altus Snyman" <[EMAIL PROTECTED]> To: "asterisk" Sent: Monday, April 18, 2005 3:29 PM Subject: [Asterisk-Users] hangs pc Good day all I installed asterisk on a pc with redhat 9 and a 4port bri eachtime a call comes in,iax,sip,pstn it just hangs the pc Top shows 75% of the cpu goes to asterisk? Any Idea why? Please Help ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dynamic Dialplan - Turn VM on/off?
Thanks Boris. I think I can follow that logic!
- Original Message -
From: "Boris Bakchiev" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Monday, April 18, 2005 4:17 PM
Subject: RE: [Asterisk-Users] Dynamic Dialplan - Turn VM on/off?
Rod,
Here is my macro for this:
[macro-sipexten]
exten => a,1,VoicemailMain(${ARG1})
exten => a,2,Hangup()
exten => s,1,DBGet(NATIMEOUT=Features/${EXTEN}/NATIMEOUT)
exten => s,2,Dial(${ARG2},${NATIMEOUT})
exten => s,3,Goto(s-${DIALSTATUS},1)
exten => s,102,Goto(s,350)
exten => s,350,SetVar(NATIMEOUT=30)
exten => s,351,Goto(s,2)
As you can see it picks it up from DB with default being 30secs if no DB
entry exist.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Rod Bacon
Sent: Monday, 18 April 2005 15:58
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Dynamic Dialplan - Turn VM on/off?
G'day. I've been working with * for some time now, but mostly from a
enterprise perspective. I've just setup my own box at home and want to
enable some more "home user" type functionality.
Does anyone have a trick to allow the dynamic modification of the
dialplan by users? I want the ability to switch voicemail on/off (or
at
least alter the timeout).
In essence, I want to simulate the act of manually turning an
answering
machine on when you leave home (for my wife).
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Re: [Asterisk-Users] IPP g723 and getting error when starting asterisk
I had the same problem with the G729 codec because I had forgotten to edit the Makefile and select the correct optimizations for my processor. - Original Message - From: "CM Rahman Jr." <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Monday, April 18, 2005 10:53 AM Subject: [Asterisk-Users] IPP g723 and getting error when starting asterisk The compilation of codec g723.1 was fine. After I have copied to /usr/lib/asterisk/modules and started the asterisk -c .. I get this below error before asterisk quit. Anybody had any idea on Intel codec 723.1 ? [codec_g723.so] => (G723.1/PCM16 (signed linear) Codec Translator, based on IPP) Illegal instruction [EMAIL PROTECTED] G723.1]# Ouch ... error while writing audio data: : Broken pipe Thanks &*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&* C.M. Rahman Jr. IT Manager CCNP, MCSE Security"Secure your self by securing your System" CompTI Security Plus Certified CCS Internet http://www.ccsi.com 13706 Research Blvd. Suite 100 Austin, TX 78750 Tel: 512-257-2274 Ex: 115 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of denon Sent: Sunday, April 17, 2005 5:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IPP g729 & x86_64 I'm curious, how are you licensing your codec? The source is open, but the codec usage licensing is not. I think you'll find that licensing it from Digium will be much simpler, not to mention their code will Just Work(tm) without any messing around. -d At 12:08 PM 4/17/2005, you wrote: Hi, I 'm using a server DL145 with AMD opteron processors, with TE410P Digium Quad-Span card. The server is running RHEL4 x86_64. And have problem to compile codec g729 from http://www.readytechnology.co.uk/open/g729/, but ipp sample speech code not problem compile with ia32 or em64t. use l_ipp_ia32_itanium_p_4_1_2 : gcc -shared -static -Xlinker -x -o bin/codec_g729.so samples/util_e.o samples/util_d.o samples/codec_g729.o api/decg729fp.o api/encg729fp.o api/owng729fp.o api/usc729fp.o -L/opt/intel/ipp41/ia32_itanium/lib -lippscmerged -lippsrmerged -lippsmerged -lippcore -lpthread -lm /usr/bin/ld: /usr/lib/gcc/x86_64-redhat-linux/3.4.3/crtbeginT.o: relocation R_X86_64_32 against `__deregister_frame_info' can not be used when making a shared object; recompile with -fPIC /usr/lib/gcc/x86_64-redhat-linux/3.4.3/crtbeginT.o: could not read symbols: Bad value collect2: ld returned 1 exit status make: *** [bin/codec_g729.so] Error 1 Iand use from l_ipp_em64t_p_4_1_2 : gcc -shared -static -Xlinker -x -o bin/codec_g729.so samples/util_e.o samples/util_d.o samples/codec_g729.o api/decg729fp.o api/encg729fp.o api/owng729fp.o api/usc729fp.o -L/opt/intel/ipp41/em64t/lib -lippscem64t -lippsrem64t -lippsem64t -lippcoreem64t -L/opt/intel/ipp41/em64t/sharedlib/linuxem64t -lguide -lpthread -lm /usr/bin/ld: /usr/lib/gcc/x86_64-redhat-linux/3.4.3/crtbeginT.o: relocation R_X86_64_32 against `__deregister_frame_info' can not be used when making a shared object; recompile with -fPIC /usr/lib/gcc/x86_64-redhat-linux/3.4.3/crtbeginT.o: could not read symbols: Bad value collect2: ld returned 1 exit status make: *** [bin/codec_g729.so] Error 1 Any thoughts? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF in outbound calls
I have an application that creates a call file and drops it into the /var/spool/asterisk/outgoing directory. This file places a call and presents a menu to the called party. This works perfectly over SIP phones but will not work via my ISDN PRI (Digium TE410P). If I dial IN to the menu, DTMF is detected but if Asterisk originates the call, DTMF is not recognised. Does anyone have any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Citrix
The citrix ICA protocol is fundamentally BAD for audio/video application in general, let alone something as time sensitive as VOIP. - Original Message - From: "Javier Godinez" <[EMAIL PROTECTED]> To: Sent: Tuesday, April 19, 2005 12:30 PM Subject: [Asterisk-Users] Citrix Has anyone out there found a VoIP client that is citrix compatible? I am connecting to a virtual machine via citrix and want to launch a citrix compatible soft phone to connect to another virtual machine running asterisk. Does anyone have a similar setup out there? Thanks, Javier ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF in outbound calls
Does the fact that I have received no responses mean (1) nobody has suffered this problem, or (2) it's a stupid question that has been solved before and the answer is freely available elsewhere on the 'net. If it's 2, please tell me so, and I'll continue to search! - Original Message ----- From: "Rod Bacon" <[EMAIL PROTECTED]> To: Sent: Tuesday, April 19, 2005 4:01 PM Subject: [Asterisk-Users] DTMF in outbound calls I have an application that creates a call file and drops it into the /var/spool/asterisk/outgoing directory. This file places a call and presents a menu to the called party. This works perfectly over SIP phones but will not work via my ISDN PRI (Digium TE410P). If I dial IN to the menu, DTMF is detected but if Asterisk originates the call, DTMF is not recognised. Does anyone have any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Citrix
Ultimately, Citrix will be a recognised player in the VOIP space. The fact that they acquired Net6 last year should be a reasonable indicator. http://www.brianmadden.com/content/content.asp?ID=277 My comments were based on my personal experience with ICA over the last 10 years, and also based on where ICA and Metaframe are right now. Indeed, there are a number of unused channels in the ICA protocol that will (potentially) facilitate the carrying of voice data with the appropriate prioritisation. These were originally added to allow such things as streaming audio and video (and one would assume, VOIP). Keep in mind that the nature of the ICA protocol has always been fairly asymmetrical. There was never really the need to send much more than keyboard/mouse data back to the server. Historical movements of Citrix into the multimedia arena have been reasonably forgettable... just google for info on the Citrix Videoframe product. - Original Message - From: "Loek Gijben" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, April 20, 2005 5:17 AM Subject: Re: [Asterisk-Users] Citrix L.S. The citrix ICA protocol is fundamentally BAD for audio/video application in general, let alone something as time sensitive as VOIP. Just stating that something is fundamentally bad without reveiling something of the fundaments might not be very helpful... To be clear: bidirectional sound support is planned for Metaframe 3.0, you cannot use a microphone with current Citrix versions (try some sound application and you will see that a sound input is missing). In theory there could be use for a Softphone running from a Citrix session. When there is plenty of speed and bandwith, and a beast of a Citrix server, there is no fundamental reason not to try it. However there are three points you must consider: 1) There will probable no setting to give priority for a realtime sound application on the server, so you must compete with all other Citrix sessions for processing power. It might work on some light loaded machines. 2) There will probably be no setting to give sound priority in the ICA channel. So for instance a print job will render your voice application useless. 3) Your softphone must keep the settings in userspace, and you will have to set each client with some non conflicting ports by hand. For a quick and dirty test I put Dante's DIAX on my Citrix server. It installs without a problem, and I could use it to retrieve my voicemail without a problem. But DIAX stores the settings in a configfile in the directory where the executable is located (you could install Diax in each one's user directory off course :) To comment on Rod Baker: there is nothing wrong with the ICA protocol as it is nothing more than a kind of wrapper around some streams (or channels). Just like any other wrapper like say an IP tunnel. It is just that there will likely be no incentive for Citrix to add any prioritation scheme for sound. I wonder what the bidirectional support will be used for. Maybe speech recognition for Office programs? - Original Message - From: "Javier Godinez" <[EMAIL PROTECTED]> To: Sent: Tuesday, April 19, 2005 12:30 PM Subject: [Asterisk-Users] Citrix Has anyone out there found a VoIP client that is citrix compatible? I am connecting to a virtual machine via citrix and want to launch a citrix compatible soft phone to connect to another virtual machine running asterisk. Does anyone have a similar setup out there? Thanks, Javier ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to create channel of type 'Zap'
You have 2 problems.
Zap/g2 is one. You are trying to dial out a non-existent Zaptel group.
Change your TRUNK variable to Zap/1-1 (or just Zap/1 will do).
Also, you are stripping the 1st number off your outgoing call. If you don't
want to do this, then change TRUNKMSD to 0.
- Original Message -
From: "Jaime Blanco" <[EMAIL PROTECTED]>
To:
Sent: Thursday, April 21, 2005 4:33 AM
Subject: [Asterisk-Users] Unable to create channel of type 'Zap'
Hi,
I just installed the asterisk and the X100P card. I can receive calls
from PSTN and it can ring on a Grandstream SIP Phone. From the SIP Phone
I can dial the demo extension on asterisk pbx. The issue is as soon as I
try to dial out 92714756 or another number I received the following
message:
*CLI> -- Executing Dial("SIP/1001-2b93", "Zap/g2/2714756") in new
stack
Apr 20 02:27:40 NOTICE[245776]: app_dial.c:536 dial_exec: Unable to create
channel of type 'Zap'
== Everyone is busy at this time
-- Executing Congestion("SIP/1001-2b93", "") in new stack
== Spawn extension (from-sip, 92714756, 2) exited non-zero on
'SIP/1001-2b93'
Zapata.conf is:
[channels]
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
echocancel=yes
echocancelwhenbridged=no
rxgain=0.0
txgain=0.0
immediate=no
context=default
signalling=fxs_ks
channel=1
extensions.conf
;
; Static extension configuration file, used by ; the pbx_config module.
This is where you configure all your ; inbound and outbound calls in
Asterisk.
;
;
; The "General" category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite ;
this file when extensions are modified. Remember that all comments ; made
in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by ; CLI
command 'save dialplan' too ; writeprotect=no
; You can include other config files, use the #include command (without
the
';')
; Note that this is different from the "include" command that includes
contexts within ; other contexts. The #include command works in all
asterisk configuration files.
;#include "filename.conf"
; The "Globals" category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
variable ; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid ; [globals]
CONSOLE=Console/dsp ; Console interface for
demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/g2; Trunk interface
TRUNKMSD=1 ; MSD digits to strip
(usually 1 or 0)
;TRUNK=IAX2/user:[EMAIL PROTECTED]
;
; Any category other than "General" and "Globals" represent ; extension
contexts, which are collections of extensions.
;
; Extension names may be numbers, letters, or combinations ; thereof. If
an extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a ; literal. In
patterns, some characters have special meanings:
;
; X - any digit from 0-9
; Z - any digit from 1-9
; N - any digit from 2-9
; [1235-9] - any digit in the brackets (in this example,
1,2,3,5,6,7,8,9)
; . - wildcard, matches anything remaining (e.g. _9011. matches
; anything starting with 9011 excluding 9011 itself)
;
; For example the extension _NXX would match normal 7 digit dialings,
; while _1NXXNXX would represent an area code plus phone number ;
preceeded by a one.
;
; Contexts contain several lines, one for each step of each ; extension,
which can take one of two forms as listed below, ; with the first form
being preferred. One may include another ; context in the current one as
well, optionally with a ; date and time. Included contexts are included
in the order ; they are listed.
;
;[context]
;exten => someexten,priority,application(arg1,arg2,...)
;exten => someexten,priority,application,arg1|arg2...
;
; Timing list for includes is
;
; |||
;
;include => daytime|9:00-17:00|mon-fri|*|* ; ; ignorepat can be used to
instruct drivers to not cancel dialtone upon ; receipt of a particular
pattern. The most commonly used example is ; of course '9' like this:
;
;ignorepat => 9
; so that dialtone remains even after dialing a 9.
;
;
; Here are the entries you need to participate in the IAXTEL ; call
routing system. Most IAXTEL numbers begin with 1-700, but ; there are
exceptions. For more information, and to sign ; up, please go to
www.gnophone.com or www.iaxtel.com ; [iaxtel700] exten =>
_91700NXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED])
[iaxprovider]
;switch => IAX2/user:[EMAIL PROTECTED]/mycontext
[trunkint]
;
; International long distance through trunk ; exten =>
_9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9011.,2,Congest
Re: [Asterisk-Users] Grandstream GXP-2000
Try searching the list. There's a thread from a few weeks back of exactly the same name. - Original Message - From: "Daniel Salama" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, April 21, 2005 4:23 AM Subject: [Asterisk-Users] Grandstream GXP-2000 Does anyone have any experience with this phone? I'm considering purchasing it but wish to hear if anyone has any experience with it. Thanks, Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Playing mp3's while recording voicemail
I need to do something similar and I was thinking about starting a conference call using an AGI script and MeetMe. Basically, one leg of the call is MP3, one is a recorder and the other is a live call. I have no idea how to implement it in code, but it sounds logical to me. - Original Message - From: "Rafal Kaniewski" <[EMAIL PROTECTED]> To: Sent: Friday, April 22, 2005 9:18 AM Subject: [Asterisk-Users] Playing mp3's while recording voicemail Anyone got a good idea on how to do this? (its for a singing down the phone thing) thanks Rafal Kaniewski Rafal#movingimagearts.com -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.10.1 - Release Date: 20/04/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX ATA's
What sort of price are they asking for a 4-port gateway? - Original Message - From: "Joseph" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, April 27, 2005 8:53 AM Subject: Re: [Asterisk-Users] IAX ATA's There is Taiwan company Soundwin that seems to me are willing to support asterisk protocol in their equipment. I was looking for 1FXO x 3-4FXS in one unit. http://www.soundwin.com/ I just exchanged few email with Sam at [EMAIL PROTECTED] and was able to convince them to add support for IAX2; they seems to me listen to the end user so I suggest some of you drop him an email and express your interest in their product if they will support asterisk protocol. --quote We have plan to implement IAX or IAX2 in our product line including 2 -8 port VoIP Gateway in Q3. Thanks your information and we would pay more attention in Asterisk community. -end quote- -- #Joseph On Tue, 2005-04-26 at 16:46 -0400, Garrett Smith wrote: Does anyone know of a quality alternative to the Digium IAXy? I have a customer experiencing numerous issues such as over heating with the older IAXy’s and the new IAXy is not yet available. Can anyone recommend an alternative? Thanks, Garrett Smith [EMAIL PROTECTED] B2 Technologies/ VoIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 (716) 250-3408 Direct (716) 630-1548 Fax (716) 903-9495 Cell AOL IM: B2sales Specializing in New and Used equipment from vendors including Cisco Systems, Juniper, Adtran, Dialogic, Lucent, Nortel, Sipura, Granstream, Snom, Mediatrix, Carrier Access, Digium, Zultys, IPDialog and more. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No Audio sent using playback cmd
What errors are you seeing at the console? The only time I've ever had this problem was because I specified the file extension in the filename. Eg. Playback(file.wav) is INCORRECT. Needs to be specified as Playback(file). Some more info may help to get your question answered! - Original Message - From: "Michael D Schelin" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, April 27, 2005 10:55 AM Subject: [Asterisk-Users] No Audio sent using playback cmd Hi All, I really need help on this. What would keep Asterisk from playing out audio files using the (Playback command) but I can play the busy tone . playtone(Congestion) ?? I have verified this with ethereal and see the audio only going one way. In to Asterisk bun nothing coming out. Because I can hear the audio with the play tone I know there is something preventing the playback cmd from working. Thanks _ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream ATA 286 problems
What firmware version are you running? My Grandstream phones and ATAs all work fine with .22 I don't use G.723 codec though. Mainly iLBC and G.729. - Original Message - From: "Anton Krall" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Tuesday, April 26, 2005 2:44 AM Subject: [Asterisk-Users] Grandstream ATA 286 problems Anobody had any problem with GS ata 286? The past few days Ive been having some problem with it, while making a call or during a call, I suddely hear a low noise like a car engine starting and then the ata dies, as if it got stuck or frozen. Anybody had these problems? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Warm standby boxes - keeping config syncronised?
This sounds like a Job for ARA. You can make your * servers read their config from a central database server - Original Message - From: "David John Walsh" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, April 27, 2005 10:05 AM Subject: [Asterisk-Users] Warm standby boxes - keeping config syncronised? Ok probably not strictly an asterisk question. I have an asterisk box, which is running some non-critical telephones in our organisation, and if it fails it fails. However comming from a telecoms background I always want to make things recoverable quickly. Since I have little budget, and down time isn't an issue my thoughts are as folllows 2 servers with 2 NIC's each, one nic for managment, one for traffic. 1 NIC on each machine has the same IP address, but only one is plugged into the network at any one time. 2 PRI's that are plugged into the machine that is live to traffic. Apart from the managment NIC having different IP addresses they are configured identically If I make a change to the in-service server, how do I automagically get the other server to take a copy of it? I'm not a linux man by trade, so if you say set up master / slave would you be kind enough to suggest an aplication and how it would be implimented. Thanks for any ideas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Linux SoftPhone with Sound Daemon Support
Does anyone know of a Linux SoftPhone that will play nicely with ESD? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Latest CVS Head Nukes Server
Has anyone experienced problems with recent CVS HEAD (as of 30th April) version of * completely crashing the PC on shutdown? (I can't see the console, because the server is located in remote data centre). The problem doesn't appear to happen all the time. Only when * has been running for a while. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_vpb Verbose Logging
Does anyone know if there is a way to turn DOWN the verbosity of the Voicetronix channel driver? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE410P on Dell 2650
Are you running vga=normal in your lilo.conf? (disable frame buffer) and running kernel WITHOUT apic and acpi support? (append='noapic acpi=off"). Making these changes, and disabling all other unused resources (to eliminiate IRQ sharing) got me to 100% consistently on a DELL 2850. -- ====== Rod Bacon - VOIP Systems Engineer Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Database of actve calls (as per astguiclient)
I have the need to maintain a pseudo-realtime database of active calls across a number of asterisk servers. The main purpose of this is in determining where to route calls (e.g. don't send calls to a server with no free lines) and also for monitoring/recirding calls. I know that astguiclient does this by telnetting into the * server management interface ever 333ms and updating a MYSQL database. Does this place much load on the * server, or the DB server? Will this sort of model scale to 30 servers, each with 120 Zap channels? -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN Clock Source
I apologise in advance if this is a silly question, as "legacy" telephone technologies are really not my forte. Is there an E1 card that can provide clock source? (E.g. Make my asterisk server look like a telco to my legacy PBX system?). What I am trying to achieve is the following: --ISDN---| Asterisk |---ISDN| Legacy PBX |-- -- ====== Rod Bacon - VOIP Systems Engineer Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outbound Caller ID on PRI
We have just had an ISDN PRI service commissioned here in AU (using Powertel as provider). I have called them and ensured that we have the ability to set Caller ID on our service to any number in our 100 number block, and I have been assured that everything is OK from their end (unlikely). Every time I drop a sample.call in to /var/spool/asterisk/outgoing with the Callerid: option set, the resultant call still appears to come from our main number. I want to make sure that it's definitely not our problem before I hassle the telco. I have debugged the PRI span, and have seen the following. Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number passed network screening (1) '0386172169' ] I am using a Zaptel Quad PRI card, and currently have an ISDN-10 service connected. Can anyone offer any suggestions? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Festival Woes
SIOD ERROR: wrong type of argument to car : wholeutt Try changing your festival.scm to the following: (Note the extra () on the 4th last line). (define (tts_textasterisk string mode) "(tts_textasterisk STRING MODE) Apply tts to STRING. This function is specifically designed for use in server mode so a single function call may synthesize the string. This function name may be added to the server safe functions." (let ((wholeutt (utt.synth (eval (list 'Utterance 'Text string) (utt.wave.resample wholeutt 8000) (utt.wave.rescale wholeutt 5) (utt.send.wave.client wholeutt))) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outbound Caller ID on PRI
Some more info on my problem that someone may be able to explain. The debug information (shown below), lists the LENGTH of the CallerID string as 14 characters, even though I'm only sending 10. I belive that this is the problem. My telco's equipment is looking for 10 characters only. Any ideas where these extra 4 characters are coming from? >Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) >Presentation: Presentation permitted, user number passed network screening (1) '0386172169' ====== Rod Bacon - VOIP Systems Engineer Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237IAXtel: 17007401708 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outbound Caller ID on PRI
>think TON = national you should send a CID of 386172169 ie, without >the leading 0 >Although I use this: >exten => s,4,SetCIDNum() >ie, just the 8 digits, which works fine... >and my zapata.conf has: >pridialplan=local I'm sure there is a setting somewhere, but I'm damned if I can find it. I have tried all possible values for prilocaldialplan (Calling number plan) but nothing makes any difference. The LEN field is always 14 (no matter what string I send). ====== Rod Bacon - VOIP Systems Engineer Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237IAXtel: 17007401708 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outbound Caller ID on PRI
Would someone mind doing an intense debug on their ISDN PRI and see what LEN (length) the calling number field is being sent? Maybe everyone is sending 14 characters, and my Telco is just fussier than most. == Rod Bacon - VOIP Systems Engineer Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600 Fax: +613 99401650 FWD: 512237 IAXtel: 17007401708 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] A hypothetical question...
I know this is casting a wide net, but If you were charged with building a large, public VOIP network with multiple PSTN gateways, the capacity to carry a lot of traffic and bill clients accurately, what pieces (brands, makes, models) would you use to assemble the solution? Assume that $$$ is not an issue. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] A hypothetical question...
Dean, which “M” would you have me read? I have been “R”ing “T”he “F”ing “M” for several months now, and have tried a number of products personally, but my very point is that it is physically impossible to test ALL PSTN gateways, ALL softswitches, ALL radius/billing solutions myself. I was counting on getting some meaningful comments from people who have positive/negative experience with various VOIP network components. Thanks to those who submitted meaningful responses… …but I’m pretty sure I’ll steer well clear of Call Mangler. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outbound Caller ID on PRI
As it turns out, it was a telco configuration problem all along. I wasted a day for nothing... - Original Message - From: "George Cohn" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, February 18, 2005 2:38 PM Subject: Re: [Asterisk-Users] Outbound Caller ID on PRI Rod Bacon wrote: Would someone mind doing an intense debug on their ISDN PRI and see what LEN (length) the calling number field is being sent? Maybe everyone is sending 14 characters, and my Telco is just fussier than most. Not asterisk related but on the Nortel Opt 81C switches that I maintain, the CLID is sent out on the PRI-ISDN span d-channels as (520) 873- which I believe is 14 characters. It shows up on my caller ID unit at home as (520) 873- which is 14 characters. The telco I am connected to is Time Warner and their CO switch is a Nortel DMS-500 running NI2 compatible software. This is in Arizona USA. George Cohn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SpanDSP - Still can't send
I have googled until blie in the face, WiKi'd until physically exhausted and searched through every Asterisk repository that I can find, all to no avail... No matter which version of SpanDSP I use, with which version of libtiff, Asterisk, ... I simply cannot send faxes. I can receive faxes *perfectly* (sending from standard fax macine on POTS) into tiff file (via Digium analogue card), but when I try to send the same file back, to the same fax machine, nothing happens. The fax rings, answers, then disconnects shortly after (fails negotiation?). I would love to enable logging in detail, but am unsure where to enable logging for SpanDSP. Can anyone point me in the most logical direction, please? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SpanDSP - Still can't send
My understanding is that this is only required when using it inside a dialplan. Eg, the extension answers, then switches to fax originator mode. For testing, I am using txfax within a sample.call file which I drop into /var/spool/asterisk/outgoing, as per latest docs. Listening at the receiving fax, the tones "sound" right, although I must confess that I don't actually speak "fax". Does anyone have any ideas about debugging SpanDSP? - Original Message - From: "Peter Svensson" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, February 23, 2005 6:28 PM Subject: Re: [Asterisk-Users] SpanDSP - Still can't send On Wed, 23 Feb 2005, Rod Bacon wrote: No matter which version of SpanDSP I use, with which version of libtiff, Asterisk, ... I simply cannot send faxes. Did you remember to add the "caller" option to txfax? Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SpanDSP - Still can't send
I did. No joy. Output from Asterisk console below. The parameter is getting through OK, but same result. -- Attempting call on Zap/1/650 for application txfax(/tmp/test.fax|caller) (Retry 1) > Channel Zap/1-1 was answered. > Lauching txfax(/tmp/test.fax|caller) on Zap/1-1 -- Hungup 'Zap/1-1' - Original Message - From: "Peter Svensson" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, February 24, 2005 8:38 AM Subject: Re: [Asterisk-Users] SpanDSP - Still can't send On Thu, 24 Feb 2005, Rod Bacon wrote: My understanding is that this is only required when using it inside a dialplan. Eg, the extension answers, then switches to fax originator mode. No. To quote the mail from Steve Underwood: "Many people have it working that way. Very few people use it for fax pickup. To change it from answerer to originator mode you just need to add the "caller" option when you call txfax." You need to tell txfax that it is to act as the initiative-taker (originator) of the fax conversation. This is orthogonal to the direction of the transfer (that is selected by choosing txfax or rxfax). SpanDSP defaults to being the terminator and not the originator unless given the "caller" parameter. Have you tried adding the "caller" option to txfax? For testing, I am using txfax within a sample.call file which I drop into /var/spool/asterisk/outgoing, as per latest docs. Listening at the receiving fax, the tones "sound" right, although I must confess that I don't actually speak "fax". ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SpanDSP - Still can't send
From my personal experience, the 'weird ideas' come from a lack of consistent documentation. The caller option, and the use thereof is not clearly explained anywhere that I can find, and examples of spandsp that are floating around the WiKi (among other places) erroneously leave this option out. In any case, I still can't send faxes... with or without the option. - Original Message - From: "Steve Underwood" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, February 24, 2005 11:02 AM Subject: Re: [Asterisk-Users] SpanDSP - Still can't send Where do these weird ideas come from? :-) Why on earth would it relate to the dial plan? What difference can you imagine the dial plan might make? If you want txfax to behave as the calling, rather than the answering, side you need to specify the "caller" option. Don't confuse caller with sender. The machine which makes a call can be either the one sending a FAX, or one picking up a FAX by polling. That is the reason the "caller" option exists. Regards, Steve Rod Bacon wrote: My understanding is that this is only required when using it inside a dialplan. Eg, the extension answers, then switches to fax originator mode. For testing, I am using txfax within a sample.call file which I drop into /var/spool/asterisk/outgoing, as per latest docs. Listening at the receiving fax, the tones "sound" right, although I must confess that I don't actually speak "fax". Does anyone have any ideas about debugging SpanDSP? - Original Message - From: "Peter Svensson" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, February 23, 2005 6:28 PM Subject: Re: [Asterisk-Users] SpanDSP - Still can't send On Wed, 23 Feb 2005, Rod Bacon wrote: No matter which version of SpanDSP I use, with which version of libtiff, Asterisk, ... I simply cannot send faxes. Did you remember to add the "caller" option to txfax? Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] multiple sip phones behind firewall
There are so many possibilities here... First, get hold of every whitepaper that you can find on NAT Traversal in SIP, so you at least understand the issue. In my case, with Grandstream phones, I set them to use STUN, and make sure that they use a dynamic port. Your ultimate solution will be dependent on your Phones and your Firewall, and whether you intend on registering the phones directly on Asterisk, or via a SIP proxy (like SER) first. - Original Message - From: "Paul P. Pongco" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, February 24, 2005 1:00 PM Subject: [Asterisk-Users] multiple sip phones behind firewall Hello List, Can you please point me to the right resources on making multiple sip phones behind a firewall w/ private address work with asterisk w/c is on a public network. I have seen STUN on the grandstream and Xtunnels on X-lite. What is most deployed by members here with similar setups? Thanks. -- Cheers, Paul P. Pongco ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SpanDSP - Still can't send
Thanks Steve, and please don't think I am criticising you or anyone else. Everyone who is working on Asterisk and related products are pure legends as far as I'm concerned... I wish I could help in some way (I'm not a programmer's backside!). - Original Message - From: "Steve Underwood" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, February 24, 2005 7:25 PM Subject: Re: [Asterisk-Users] SpanDSP - Still can't send Fair point. The FAX apps used to display a note about this when you checked their parameters. That seems to have disappeared from the source code of both rxfax and txfax. I just put it back in. Regards, Steve Rod Bacon wrote: From my personal experience, the 'weird ideas' come from a lack of consistent documentation. The caller option, and the use thereof is not clearly explained anywhere that I can find, and examples of spandsp that are floating around the WiKi (among other places) erroneously leave this option out. In any case, I still can't send faxes... with or without the option. - Original Message - From: "Steve Underwood" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, February 24, 2005 11:02 AM Subject: Re: [Asterisk-Users] SpanDSP - Still can't send Where do these weird ideas come from? :-) Why on earth would it relate to the dial plan? What difference can you imagine the dial plan might make? If you want txfax to behave as the calling, rather than the answering, side you need to specify the "caller" option. Don't confuse caller with sender. The machine which makes a call can be either the one sending a FAX, or one picking up a FAX by polling. That is the reason the "caller" option exists. Regards, Steve Rod Bacon wrote: My understanding is that this is only required when using it inside a dialplan. Eg, the extension answers, then switches to fax originator mode. For testing, I am using txfax within a sample.call file which I drop into /var/spool/asterisk/outgoing, as per latest docs. Listening at the receiving fax, the tones "sound" right, although I must confess that I don't actually speak "fax". Does anyone have any ideas about debugging SpanDSP? - Original Message - From: "Peter Svensson" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, February 23, 2005 6:28 PM Subject: Re: [Asterisk-Users] SpanDSP - Still can't send On Wed, 23 Feb 2005, Rod Bacon wrote: No matter which version of SpanDSP I use, with which version of libtiff, Asterisk, ... I simply cannot send faxes. Did you remember to add the "caller" option to txfax? Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Static on TDM Zaptel FXO
Make sure you have disabled framebuffer, apic and acpi. -- == Rod Bacon - VOIP Systems Engineer Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] more than one company hosting their PBX on the same machine?
Sigh... read the wiki. Search the lists. This has been answered at least fifteen times. You don't need multiple instances of *, just set up your dialplan properly. Hint: Contexts are the key. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Which H.323 for Stable?
I'm new to H.323 and I have noticed that there are two separate channel drivers for * available - the inbuilt one, and oh-323. I had trouble compiling oh-323 with the current cvs stable, so I tried the inbiult one (with specifiec recommended versions of openh323 and pwlib). It compiled cleanly but I am told that it is not recommended (unstable?). Can someone with first-hand * H.323 experience offer any meaningful advice as to which way I _must_ proceed? This is for a live, busy, deployed environment. H.323 will be used to connect to an upstream provider (possibly CISCO gear?). Thanks in advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems trying to compile H323 from CVS-STABLE
Tony, I have managed to compile both versions (on separate servers, obviously), and have them working. My question is specifically related to "which one do I choose?". To get the "internal" version working, I used the oldest versions of the libraries that I could find. Specifically, the 28th Aug 2003 builds from voxgratia.org (PWLib 1.5.3 and OpenH323 1.12.3) When I first loaded it, I DID get output from "h.323 show codecs"... now, strangely, it's empty. Also, "h.323 show tokens" reveals nothing. Call establishment, audio, call teardown all seem OK, but all calls seem to be in ULAW, no matter what I specify. The oh-323 channel seems OK, but doesn't like to play with certain codecs. I'm also concerned with the file handle situation you described. How can I debug this to see what is happening? (I'm not a programmer, so be gentle!) -- == Rod Bacon - VOIP Systems Engineer Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Analog Telephone Adapter
An IBM sales rep once told me... I can give you RELIABLE, FAST and CHEAP... any two of them at once. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] International Caller ID?
We have antiquated caller ID schemes here in Australia. We barely support numbers from other local carriers, let alone OS ones. Certainly no names either. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CRM integration (was RE: CallerID)
This sounds remarkably like an IM problem We're in the process of building a CRM frontend that uses Jabber as the IM mechanism. The Asterisk server sends the URL via Jabber (PCs authenticated as extension number). The Jabber client (custom, written in Flash) receives the URL and automagically follows it. Michiel van Baak wrote: On 20:31, Sat 28 May 05, Gavin Hamill wrote: On Saturday 28 May 2005 20:21, Rusty Shackleford wrote: D'oh! I had misread the PP's statement and assumed he meant a "bareback" browser window. You are, of course, quite right. A Java app could handle this, but we are still left with the issue of having to install SOMETHING, even if it is a small Java app, on the client to make this work. What about this 'Ajax' stuff that's terribly trendy right now? It'd be a horrible polling implementation, but you could use a javascript Timer object to fire an XmlHTTPRequest every couple of seconds to check for new callerID at the IP address of the current browser? Cheers, Gavin. I dont do it with Ajax, but with my own written xmlhttprequest javascript. Did you check out my tar.gz file ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Inject Audio into Existing Call
Other than using a conference, does anyone know of a way to "inject" audio into a live call between two parties? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Box as a Router, Firewall and DHCP Server
On my * box at home (a dual PIII 1.2Ghz with 512Mb RAM), I'm running * (2 single-port FXO cards and SIP/IAX upstreams), MythTV (home theatre SW), file & print services and other ancillary services. I have enough CPU grunt to decode video (watch DivX) and talk on the phone (inc transcoding). * on it's own is reasonably light on resources. Go for it! ====== Rod Bacon - VOIP Systems Engineer Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 == Samy Antoun wrote: Hi, I'm planning to get my Asterisk box out of the LAN, get rid of my router and make the box acts as a Router, Firewall, DHCP Server (with Shorewall). I'll do that to be able to use some SIP clients remotely. Does anyone doing the same with the Asterisk box, is it a good idea, is there any other solution for the SIP emote Clients. Regards. __ Discover Yahoo! Stay in touch with email, IM, photo sharing and more. Check it out! http://discover.yahoo.com/stayintouch.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ztcfg server crash
I was wondering if anyone had experienced the following with asterisk stable. After a period of time (can vary), If I stop asterisk and try to run ztcfg -v to reinitialise my quad e1 card, the server will lock up. Sometimes it's a complete lockup, where it won't even return pings, other times it seems to be "partially screwed". -- ========== Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ztcfg server crash
I am running Debian Sarge with a custom 2.6.11 kernel. I'll try building another kernel and recompiling the zaptel stuff. Jason Walker wrote: What OS/distro are you running? I experienced the same on Gentoo with the 2.6.xxx kernel. Switched to FC1 (2.4.xxx kernel) with the 1.0.7 CVS and have not had any issues. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon Sent: Monday, June 13, 2005 7:31 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ztcfg server crash I was wondering if anyone had experienced the following with asterisk stable. After a period of time (can vary), If I stop asterisk and try to run ztcfg -v to reinitialise my quad e1 card, the server will lock up. Sometimes it's a complete lockup, where it won't even return pings, other times it seems to be "partially screwed". ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ztcfg server crash
Thanks for the info. Sergio Serrano wrote: Before change OS try to do next steps: first, stop asterisk. Second, you must do "ztcfg -s" to shutdown span. Unload modules, load modules if you need and do ztcfg -vv again. Start asterisk Regards Srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Jason Walker Enviado el: martes, 14 de junio de 2005 6:07 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Asunto: RE: [Asterisk-Users] ztcfg server crash I tried to get * stable on a 2.6xxx kernel for about 2 weeks. Then tried it out on a FC1 2.4.xxx kernel and found none of the issues. I am sure others have had success with > 2.4.xxx, but I gave up;) BTW - I was using a TE110P and then a TE405P card for the zaptel install. Both were setup as T1s not E1s. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon Sent: Monday, June 13, 2005 7:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ztcfg server crash I am running Debian Sarge with a custom 2.6.11 kernel. I'll try building another kernel and recompiling the zaptel stuff. Jason Walker wrote: What OS/distro are you running? I experienced the same on Gentoo with the 2.6.xxx kernel. Switched to FC1 (2.4.xxx kernel) with the 1.0.7 CVS and have not had any issues. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon Sent: Monday, June 13, 2005 7:31 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ztcfg server crash I was wondering if anyone had experienced the following with asterisk stable. After a period of time (can vary), If I stop asterisk and try to run ztcfg -v to reinitialise my quad e1 card, the server will lock up. Sometimes it's a complete lockup, where it won't even return pings, other times it seems to be "partially screwed". ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OH323 Packetization
Forgive this (possibly) silly question, but my upstream provider requires a packetization of 20ms. Using asterisk-oh323, I can set the "number of frames per RTP packet". How does this equate to packetization in ms? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OH323 Packetization
ok. I've worked out that G.711 is 1ms of audio per frame... what about G.729? Rod Bacon wrote: Forgive this (possibly) silly question, but my upstream provider requires a packetization of 20ms. Using asterisk-oh323, I can set the "number of frames per RTP packet". How does this equate to packetization in ms? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OH323 Packetization
I answered my own silly question. 10ms. If anyone needs a working OH323 config for Comindico (SPT) in Australia, please mail me (G.729 and G.711). Rod Bacon wrote: ok. I've worked out that G.711 is 1ms of audio per frame... what about G.729? Rod Bacon wrote: Forgive this (possibly) silly question, but my upstream provider requires a packetization of 20ms. Using asterisk-oh323, I can set the "number of frames per RTP packet". How does this equate to packetization in ms? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream ATA Toasted
A BETA firmware upgrade toasted my ATA286. It now has limited operations. It will get an IP address via DHCP and register to the last configured SIP server, but the web interface is gone as is the voice config menu. Apart from registration, there doesn't appear to be any other SIP functionality. An Ethereal dump does not show the device trying to grab a new firmware via tftp on bootup, so this is not an option either. Can it be fixed, or is it now rubbish? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream ATA Toasted
This is not an option for me, as the IVR menu is nuked as well... Luki wrote: A BETA firmware upgrade toasted my ATA286. It now has limited operations. Happened to me too... looked mostly dead, but not quite. Try a complete hardware reset. See section 8 on http://www.grandstream.com/user_manuals/HandyTone.pdf --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Scratchy audio on Bridged PRI Calls
I have a number of servers with TE405P cards. The servers are DELL 1850's (which I _NOW_ see are listed on the digium "not recommended page" because of the ethernet interface). The problem I have is only during bridged calls. If I place a call into a service hosted on the box, or out to a VOIP phone, audio is crystal clear. If place a call "through" the box (a bridged PSTN call) the calling party hears some form of distortion when the other party speaks. Almost like a buzzing/crackling sound. I have been right through the interrupt sharing issue (disabled ACPI, APIC, Hyperthreading, Frame Buffer, etc. etc.) and am getting good results in zttest. I see NO IRQ misses or any other errors at the console. Does anyone have any other ideas? -- ========== Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Premptible Linux Kernel
Can anyone tell me if Asterisk would speficically benefit from the reduced latency of a preemptible Linux Kernel? I know it was recommended against in the early days, but I'm wondering if there are any updated opinions? -- ====== Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel HEAD with * Stable?
Will the CVS HEAD version of the Zaptel drivers work with the STABLE branch of *? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ee1000 Ethernet in Dell 1850
Digium's site now lists the Dell 1850 as a potential problem server, as it uses the intel ee1000 Ethernet chipset (as do a majority of servers in the market!). To my knowledge, ALL dell servers with Gigabit interfaces now use the same chipset. Does this mean the Digium cards can't be used in Dell servers unless you disable the onboard ethernet? I don't want to disable the onboard interface, as I use the IPMI management facility for lights-out management. I have a 2850 that doesn't have any audio problems (the reason that I contacted Digium in the first place), so I'm wondering if Digium are simply guessing at problems. Does anyone know anything specific about the supposed incompatibilities with the ee1000 kernel module? There seems to be an ever-growing list of situations where you can't use the Digium cards. This is a concern to me. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel card AND Ztdummy together?
I had a weird (unforeseen) situation today. We have a remote office with an * server and ISDN 10 service. We connect to each other over an IAX trunk with G729. Today, some of Sydney experienced a power surge which knocked out their ISDN services. Without a clock source on their PRI card, my IAX calls to them resulted in one-way audio (they could hear me, but I not them). Is it possible to load *both* the relevant card driver *and* ztdummy to guard against this occurrance? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel card AND Ztdummy together?
It's a Digium single-port job. No other timing sources aviailable (the * box IS the pbx). qrss wrote: What kind of card are they using? Is there only 1 telco circuit? If so, then I'm thinking their card should have detected the loss of service and switched to it's internal clock. Do they have a secondary clock source available across another circuit? Perhaps a tie line to a pbx that can be configured as a secondary? -Original Message- From: Rod Bacon Sent: Thu, June 23, 2005 12:03 am I had a weird (unforeseen) situation today. We have a remote office with an * server and ISDN 10 service. We connect to each other over an IAX trunk with G729. Today, some of Sydney experienced a power surge which knocked out their ISDN services. Without a clock source on their PRI card, my IAX calls to them resulted in one-way audio (they could hear me, but I not them). Is it possible to load *both* the relevant card driver *and* ztdummy to guard against this occurrance? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel card AND Ztdummy together?
In case anyone is interested, loading Ztdummy AND a card driver at the same time will result in unpredictable timing issues. We heard intermittent echo/feedback on PRI channels. Rod Bacon wrote: I had a weird (unforeseen) situation today. We have a remote office with an * server and ISDN 10 service. We connect to each other over an IAX trunk with G729. Today, some of Sydney experienced a power surge which knocked out their ISDN services. Without a clock source on their PRI card, my IAX calls to them resulted in one-way audio (they could hear me, but I not them). Is it possible to load *both* the relevant card driver *and* ztdummy to guard against this occurrance? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DISA and a long delay; ideas?
I agree. The following commands may also be of use...
.
.
exten => s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten => s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds
.
.
- Original Message -
From: "Greg Hill" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Monday, February 28, 2005 8:22 AM
Subject: RE: [Asterisk-Users] DISA and a long delay; ideas?
On Sun, 27 Feb 2005, C. Tomlinson wrote:
I have just setup a DISA setup whereby people can dial in, authenticate,
are
given a dialtone and can then call out.
Everything works however there is a 10 second delay after the user enters
the number and presses #, until the system does anything.
Here is the relevant section from my extensions.conf:
[dialtone]
exten => s,1,Authenticate(1234)
exten => s,2,DISA(no-password|dialtone_outgoing)
[dialtone_outgoing]
exten => _01.,1,Dial(${OUTGOING}/44${EXTEN:1},30,L(6:3:1))
exten => _07.,1,Playback(pbx-invalid)
HOWEVER there is a 10 second delay between the dialing (followed by #)
and
the system doing anything.
My first guess would be digit timeouts. Your patterns are _01. and _07..
These don't give asterisk any hints about how many digits to expect, so
its only choice is to wait for the maximum digit timeout period to be sure
that it doesn't make a decision early before you've entered all your
digits.
The "best" thing (in my view) would be to completely specify the digit
patterns you want users to be able to use. This gives you the opportunity
to control which numbers may be called and which may not, and it also
gives asterisk hints about what kinds of digit patterns it should expect.
These hints allow it to make faster decisions about whether a digit
pattern is complete and/or valid. An alternative would be to use the
DigitTimeout application to set a lower timeout period.
Greg
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Re: [Asterisk-Users] Grandest Free Softphone
I've been playing with a variety of them over the past month. The 'candidates' I've got down to are X-Lite, Firefly and SJphone. They all have strengths and weaknesses, and all behave differently behind my firewall (STUN client differences?). So far, I am happiest with the performance of X-Lite. It also has the most detailed configuration options. - Original Message - From: "Anton Krall" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Monday, February 28, 2005 5:57 AM Subject: [Asterisk-Users] Grandest Free Softphone Guys.. which free softphone is the best,grandest,most recommended one out there? based on your own experiences.. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] music on hold trouble
I too am having the same problem with CVS from last night. From my debugging, * never attempts to start MOH. Anyone else found this? - Original Message - From: Krystian Filiks To: asterisk-users@lists.digium.com Sent: Monday, February 28, 2005 1:46 PM Subject: [Asterisk-Users] music on hold trouble Hi All=DIV> =DIV>I seem to have a small problem with the =usic on =old button on SJPhone. =DIV> =DIV>I have 2 asterisk installations the Rapid =istribution and one from the latest CVS. =DIV> =DIV>On the rapid dist when I press the =usic on hold =utton on my SJPhone I get music on hold. =DIV> =DIV>When I do the same I get no music on =old just =ilence. =DIV>I create extension like this exten =3D> =111,1,MusicOnHold(Default), and when I dial it then I hear music, so =usic on =old works but the hold button do not. =DIV> =DIV>Can anyone help with this? =DIV> is this a bug in CVS? =DIV> =DIV> =DIV>here are debugs from both installs (1 =orking and 1 =ot working): =DIV> =DIV>** WORKING =*** =DIV>Sip read:INVITE =ip:[EMAIL PROTECTED] =IP/2.0l: 214m: i: [EMAIL PROTECTED]c: =pplication/sdpMax-Forwards: 70CSeq: 13 INVITEf: =lt;sip:[EMAIL PROTECTED]:2841>;tag=41280171719448t: =lt;sip:[EMAIL PROTECTED]>;tag=as7cf27066User-Agent: =JLabs-SJphone/1.30.252v: SIP/2.0/UDP =92.168.1.111;rport;branch=z9hG4bKc0a8016f0131c9b142227b69594d0=78 =P>v=0o=- 3318544820 3318544833 IN =P4 =92.168.1.111s=SJphonec=IN IP4 0.0.0.0t=0 =a=direction:activem=audio 16394 RTP/AVP 3 =01a=rtpmap:3 =SM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 =-11,16 =P>11 headers, 10 linesUsing latest =equest as basis =equestSending to 192.168.1.111 : 5060 (NAT)Found audio format =NKNFound audio format UNKNFound description format GSMFound =escription format telephone-eventCapabilities: us - 6, them - 2/0, =ombined = 2Non-codec capabilities: us - 1, them - 1, combined - 1We're =t xxx.xxx.xxx.xxx port 14276Answering with preferred =apability =Answering with non-codec capability 1Reliably Transmitting =NAT):SIP/2.0 200 OKVia: SIP/2.0/UDP =92.168.1.111;rport;branch=z9hG4bKc0a8016f0131c9b142227b69594d0=78;received=xxx.xxx.xxx.xxxFrom: =lt;sip:[EMAIL PROTECTED]:2841>;tag=41280171719448To: =lt;sip:[EMAIL PROTECTED]>;tag=as7cf27066Call-ID: [EMAIL PROTECTED]CSeq: =3 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, =PTIONS, =YE, REFERContact: =lt;sip:[EMAIL PROTECTED]>Content-Type: =pplication/sdpContent-Length: 219 =P>v=0o=root 17002 17015 IN IP4 =xx.xxx.xxx.xxxs=sessionc=IN IP4 195.216.65.216t=0 =m=audio =4276 RTP/AVP 3 101a=rtpmap:3 GSM/8000a=rtpmap:101 =elephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - = =P> to =xx.xxx.xxx.xxx:2841 =P> =P>*** NOT WORKING =*** =P>Sip read:INVITE sip:[EMAIL PROTECTED] =IP/2.0l: 214m: i: [EMAIL PROTECTED]c: =pplication/sdpMax-Forwards: 70CSeq: 1 INVITEf: =lt;sip:[EMAIL PROTECTED]:5060>;tag=41308811925234t: =lt;sip:[EMAIL PROTECTED]>;tag=as463b04a6User-Agent: =JLabs-SJphone/1.30.252v: SIP/2.0/UDP =92.168.1.111;rport;branch=z9hG4bKc0a8016f0131c9b142227c5f64820=c8 =P>v=0o=- 3318545106 3318545107 IN =P4 =92.168.1.111s=SJphonec=IN IP4 0.0.0.0t=0 =a=direction:activem=audio 16400 RTP/AVP 3 =01a=rtpmap:3 =SM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 =-11,16 =P>11 headers, 10 linesUsing latest =equest as basis =equestSending to 192.168.1.111 : 5060 (NAT)We're at =92.168.1.20 port =8336Answering/Requesting with root capability =x4 =ulaw)Answering with preferred capability 0x2 (gsm)Answering =ith =on-codec capability 0x1 (telephone-event)Reliably =ransmitting =NAT):SIP/2.0 200 OKVia: SIP/2.0/UDP =92.168.1.111;branch=z9hG4bKc0a8016f0131c9b142227c5f648200c8;re=eived=192.168.1.111;rport=5060From: =lt;sip:[EMAIL PROTECTED]:5060>;tag=41308811925234To: =lt;sip:[EMAIL PROTECTED]>;tag=as463b04a6Call-ID: [EMAIL PROTECTED]CSeq: = INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, =PTIONS, =YE, REFERContact: Content-Type: =pplication/sdpContent-Length: 241 =P>v=0o=root 12791 12793 IN IP4 =92.168.1.111s=sessionc=IN IP4 192.168.1.111t=0 =m=audio 16398 =TP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 =SM/8000a=rtpmap:101 =elephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - = =P> to 192.168.1.111:5060 =P>Thanks =P>KF ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-user ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/list
