[asterisk-users] iax softphones - not reconnecting
Hi together, I try to find a softphone (freeware) solution for Windows 32, that works without problems ... Right now I use iaxcomm wich was best, of the ones I tried. But I have one problem with it. When I turn on qualify, it will not connect to the asterisk. This is also documented and normal behaveor. But if I turn of qualify, iaxcomm does not reconnect to the server, when the server got restartet or the connection got lost. So I do not know, how I can get this problem solved. Turning on qualify would get it reconnecting, but after some time it does disconnect with the note: is now unreachable. is there any other option I can activate to get asterisk or iaxcomm reconnecting? Or does anybody know any other softphone for windows, that is freeware and maybe also brandable? hope anybody can help me to get this running... Thank you very much, Martin -- _ -- 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] iax client for symbian s60
Hi all, I searched for a long time and know that here this question also was asked in the past, but ... Is there any iax client for s60 now? Or still no client available? There are so many people asking for it, but nobody seems to get it done :-( cheers, Martin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Data transfer
That's not exactly true. Asterisk merely requires that a call be up in order to pass text messages. It does not, however, allow text messages to be passed stateless. Thanks for the answer, I testet it and it works for connected calls. But I have to send data even when the devices are in different conferences, so this will not work for us. Thorsten Stoffregen Sackwaldstr. 25 31061 Alfeld Tel: +49 5181 5191 Mobil: +49 173 6404335 Fax: +49 5181 807993 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime Queue not work in 1.6.2.1
All your agents have paused=1. They will not receive calls while they are paused. Håkon On Thu, Jan 28, 2010 at 3:23 AM, Zhang Shukun bit...@gmail.com wrote: 2010/1/28 Carlos Chavez cur...@telecomabmex.com: On Wed, 2010-01-27 at 10:27 +0800, Zhang Shukun wrote: hi,all i have just upgrade from 1.4.28 to 1.6.2.1. all works fine now except realtime queue. it seems queue_table works fine, but queue_member_queue not work, the two tables works fine when in 1.4.28. is that something changed related to realtime queue configuration? more detail about two table definition and data stored in , please see: http://pastebin.com/m33f9539e the extconfig.conf file, please see: http://pastebin.com/m2008ced1 and the res_mysql.conf file: http://pastebin.com/m27d3fdc5 Could you tell me what's wrong with me ? Thanks! How do your agents log into the system? Thanks! i don't want to use agents member to login to system. i just want to set static SIP peers in the queue and they all can work according to the strategy when have call to the queue.just like follows: mysql select * from queue_table; +--+---+-+ | name | beginworktime | endworktime | +--+---+-+ | 950401234561 | 09:30:00 | 17:30:00| +--+---+-+ 3 rows in set (0.00 sec) mysql select * from queue_member_table; +--++--+---+-++ | uniqueid | membername | queue_name | interface | penalty | paused | +--++--+---+-++ | 18 | Zhang Shukun | 950401234561 | SIP/1001 | 0 | 1 | | 19 | Li Aiwei | 950401234561 | SIP/1002 | 0 | 1 | | 20 | Zhang Jianming | 950401234561 | SIP/1003 | 0 | 1 | +--++--+---+-++ 3 rows in set (0.00 sec) in above two table. queue:950401234561 have three queue members: SIP/1001 , SIP/1002 , SIP/1003 when Queue(950401234561) app is invoked, all three queue members will ring at the same time by default strategy(ringall). my problem now use asterisk 1.6.2.1 is : when Queue(950401234561) app is running, i can here music on hold, but none of my sip phones(SIP/1001 , SIP/1002 , SIP/1003) will ring, is that in asterisk 1.6.2.1, it's not support static realtime queue member any more? If you were using agentcallbacklogin that was deprecated and does not exist in version 1.6 of Asterisk. The queue_member_table was used by agentcallbacklogin or the agentlogin commands. With Asterisk 1.6 you are supposed to be using dynamic agents so there is no purpose for that table. That is what may be wrong with Asterisk. What is wrong with you is a very different question ;) -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- 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 -- Best regards, Sucan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Data transfer
Hi! That's not exactly true. Asterisk merely requires that a call be up in order to pass text messages. It does not, however, allow text messages to be passed stateless. Thanks for the answer, I testet it and it works for connected calls. But I have to send data even when the devices are in different conferences, so this will not work for us. Consider to use sipsak instead, or look at a SIP proxy then (Kamalio, OpenSIPS) possibly combined with/in front of Asterisk. By the way, Asterisk writes the contents of the SIP message to the log, so at least there it is accessible. And since Asterisk is open source you can extend it as needed, I think there would be quite some interest here in slightly better testmessage features (ref. inbound SMS). Actually I wonder if chan_mobile has a better way to handle SMS. Philipp -- \\\|/// | ~ ~ | (- 0 0 -) +--oOOo-(_)-oOOo--+ | Philipp von Klitzing | | klitz...@pool.informatik.rwth-aachen.de | | Friesenstr.3, D-52062 Aachen, | |Tel/Fax: +49-241-4013340 | +-.oooO-( )--+ ( ) ) / \ ((_/ \_) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linux-based hard phones?
On 28 Jan 2010, at 02:32, Ken D'Ambrosio wrote: Just wondering if there are any Linux-based hard phones out there -- if so, it'd be neat to see if I couldn't take advantage of the underlying OS. Snom.. Cisco/Linkysys SPA.. None of them are that easy to 'take advantage' of though. S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime Queue not work in 1.6.2.1
2010/1/28 Håkon Nessjøen haa...@avelia.no: All your agents have paused=1. They will not receive calls while they are paused. Solved Thanks very much! Håkon On Thu, Jan 28, 2010 at 3:23 AM, Zhang Shukun bit...@gmail.com wrote: 2010/1/28 Carlos Chavez cur...@telecomabmex.com: On Wed, 2010-01-27 at 10:27 +0800, Zhang Shukun wrote: hi,all i have just upgrade from 1.4.28 to 1.6.2.1. all works fine now except realtime queue. it seems queue_table works fine, but queue_member_queue not work, the two tables works fine when in 1.4.28. is that something changed related to realtime queue configuration? more detail about two table definition and data stored in , please see: http://pastebin.com/m33f9539e the extconfig.conf file, please see: http://pastebin.com/m2008ced1 and the res_mysql.conf file: http://pastebin.com/m27d3fdc5 Could you tell me what's wrong with me ? Thanks! How do your agents log into the system? Thanks! i don't want to use agents member to login to system. i just want to set static SIP peers in the queue and they all can work according to the strategy when have call to the queue.just like follows: mysql select * from queue_table; +--+---+-+ | name | beginworktime | endworktime | +--+---+-+ | 950401234561 | 09:30:00 | 17:30:00 | +--+---+-+ 3 rows in set (0.00 sec) mysql select * from queue_member_table; +--++--+---+-++ | uniqueid | membername | queue_name | interface | penalty | paused | +--++--+---+-++ | 18 | Zhang Shukun | 950401234561 | SIP/1001 | 0 | 1 | | 19 | Li Aiwei | 950401234561 | SIP/1002 | 0 | 1 | | 20 | Zhang Jianming | 950401234561 | SIP/1003 | 0 | 1 | +--++--+---+-++ 3 rows in set (0.00 sec) in above two table. queue:950401234561 have three queue members: SIP/1001 , SIP/1002 , SIP/1003 when Queue(950401234561) app is invoked, all three queue members will ring at the same time by default strategy(ringall). my problem now use asterisk 1.6.2.1 is : when Queue(950401234561) app is running, i can here music on hold, but none of my sip phones(SIP/1001 , SIP/1002 , SIP/1003) will ring, is that in asterisk 1.6.2.1, it's not support static realtime queue member any more? If you were using agentcallbacklogin that was deprecated and does not exist in version 1.6 of Asterisk. The queue_member_table was used by agentcallbacklogin or the agentlogin commands. With Asterisk 1.6 you are supposed to be using dynamic agents so there is no purpose for that table. That is what may be wrong with Asterisk. What is wrong with you is a very different question ;) -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- 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 -- Best regards, Sucan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Sucan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linux-based hard phones?
Ken D'Ambrosio wrote: Just wondering if there are any Linux-based hard phones out there -- if so, it'd be neat to see if I couldn't take advantage of the underlying OS. Thanks, -Ken Snom phones use Linux Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Database
Hello
I am trying to attach a database to asterisk , can anyone help me?
in extconfig.conf sipusers = mysql,general,sip
in res_mysql.conf [general]
dbhost = 192.168.50.125
dbname = asterisk
dbuser = root
dbpass = ahmed
dbport = 3306
dbsock = /tmp/mysql.sock
i created a table in MySql
CREATE TABLE `sip` (
`name` varchar(40) NOT NULL default '',`username` varchar(40) default
'',`typee` varchar(6) NOT NULL default '',`secret` varchar(40) default '',
`context` varchar(40) NOT NULL default '',
`host` varchar(31) NOT NULL default 'dynamic',
PRIMARY KEY (`name`)
) TYPE=MyISAM
I insreted a data which is insert into sip values
('555','555','peer','1234','555','dynamic')
but i couldn't register from X-lite because Asterisk doesn't see this peer
please help me urgent
--
Ahmed Magdy Mahmoud
--
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Re: [asterisk-users] iax softphones - not reconnecting
Hi, I've used ZoiPer with our Asterisk server and not had any problems. It's quite a basic-looking client but gets the job done. I believe DIAX is another option you could try. Stuart From: Asterisk - thinking:systems aster...@tsy.at To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thu, 28 January, 2010 10:12:11 Subject: [asterisk-users] iax softphones - not reconnecting Hi together, I try to find a softphone (freeware) solution for Windows 32, that works without problems ... Right now I use iaxcomm wich was best, of the ones I tried. But I have one problem with it. When I turn on qualify, it will not connect to the asterisk. This is also documented and normal behaveor. But if I turn of qualify, iaxcomm does not reconnect to the server, when the server got restartet or the connection got lost. So I do not know, how I can get this problem solved. Turning on qualify would get it reconnecting, but after some time it does disconnect with the note: is now unreachable. is there any other option I can activate to get asterisk or iaxcomm reconnecting? Or does anybody know any other softphone for windows, that is freeware and maybe also brandable? hope anybody can help me to get this running... Thank you very much, Martin -- _ -- 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] Inserting white noise / music / sound file into mixmonitor
A week or so ago, I explained that we need to blank our call recording when some sensitive information like credit cards where being discussed. With the lists help, I managed to find the pause/ unpause monitor commands. That works great. However (there is always a however), what that now means is that the length of the call does not match the length of the call recording, so adding stuff like this happened at 11:04 into the call now is out by the length of time of the pause :( I was wondering if it was possible to replace the voice on either leg with a sound file or something, but only in mixmonitor, as we obviously need to hear the person talking in order to take the details. Julian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Data transfer
By the way, Asterisk writes the contents of the SIP message to the log, so at least there it is accessible. And since Asterisk is open source you can extend it as needed, I think there would be quite some interest here in slightly better testmessage features (ref. inbound SMS). Ok I found a quick and dirty way to grab the messages. In chan_sip.c Asterisk drops the message: ast_log(LOG_WARNING,Received message to %s from %s, dropped it...\n Content-Type:%s\n Message: %s\n, get_header(req,To), get_header(req,From), content_type, buf); transmit_response(p, 405 Method Not Allowed, req); /* Good enough, or? */ So I changed the response to: transmit_response(p, 202 Accepted, req); and send the message to the AMI: manager_event(EVENT_FLAG_CALL, MessageReceived, From: %s\r\nTo: %s\r\nContent-Type: %s\r\nMessage: %s\r\n, get_header(req,From), get_header(req,To), content_type, buf); Its a quick way for me to get the messages, so I can go on and put a prototype together ;-) And do some further testing Thorsten Stoffregen -- _ -- 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] Fw: OfficeSIP Softphone
Hello, Could anyone help to review the log and issue? Where I could post asterisk bugreport? I could help with testing if someone try to fix this error. Best regards, Vitali Fomine - Original Message - From: Vitali Fomine supp...@officesip.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 22, 2010 3:33 PM Subject: Re: [asterisk-users] OfficeSIP Softphone Hello, I would like to see this as well, from an Asterisk CLI log perspective with sip debug turned on. The .log file for login and invite is attached, I have use asterisk -vr command. Is it correct? Yes, here is two INVITEs (I have missed first invite before), but the server respond 401 on first invite and softphone send ACK. Here is softphone log. If Asterisk receives the ACK *after* the second INVITE I understand it. The softphone uses single tcp connection, so messages must arrive in same order as them was sent. Best regards, Vitali Fomine -- _ -- 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 login-invite.log Description: Binary data -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fw: OfficeSIP Softphone
You are running an Asterisk version for SIP TCP ? your SIP UA seems talking SIP over TCP Via: SIP/2.0/TCP 192.168.1.15:56298 Max-Forwards: 70 From: sip:5...@trixbox1.local;tag=2baacde98c;epid=aa3c1b27a7 To: sip:mrasloc.trixbox1.lo...@trixbox1.local Call-ID: 28a90e7402da49159f343be9bc82b4d0 CSeq: 1 SERVICE Contact: sip:5...@trixbox1.local :56298;maddr=192.168.1.15;transport=tcp;proxy=replace;+sip.instance=urn:uuid:1ADF8582-5BD5-531A-BC2A-C76FECED0C4E User-Agent: UCCAPI/2.0.6362.67 Authorization: Digest username=56, realm=asterisk, algorithm=MD5, uri=sip:mrasloc.trixbox1.lo...@trixbox1.local, nonce=36662fdf, response=d6f90f263010891a42b3f7d46113796a Content-Type: application/msrtc-media-relay-auth+xml Content-Length: 395 -- -- Adrià Vidal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fw: OfficeSIP Softphone
Hello, Yes, unfortunately, the sip client lib does not support udp. Best regards, Vitali Fomine - Original Message - From: Adrià Vidal To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, January 28, 2010 3:49 PM Subject: Re: [asterisk-users] Fw: OfficeSIP Softphone You are running an Asterisk version for SIP TCP ? your SIP UA seems talking SIP over TCP Via: SIP/2.0/TCP 192.168.1.15:56298 Max-Forwards: 70 From: sip:5...@trixbox1.local;tag=2baacde98c;epid=aa3c1b27a7 To: sip:mrasloc.trixbox1.lo...@trixbox1.local Call-ID: 28a90e7402da49159f343be9bc82b4d0 CSeq: 1 SERVICE Contact: sip:5...@trixbox1.local:56298;maddr=192.168.1.15;transport=tcp;proxy=replace;+sip.instance=urn:uuid:1ADF8582-5BD5-531A-BC2A-C76FECED0C4E User-Agent: UCCAPI/2.0.6362.67 Authorization: Digest username=56, realm=asterisk, algorithm=MD5, uri=sip:mrasloc.trixbox1.lo...@trixbox1.local, nonce=36662fdf, response=d6f90f263010891a42b3f7d46113796a Content-Type: application/msrtc-media-relay-auth+xml Content-Length: 395 -- -- Adrià Vidal -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iax softphones - not reconnecting
Stuart, thank you... Zoiper is not accessible for screenreaders right now, so we cannot use this at this time. I tried diax v0.910f This would be accessible, but it does not save account informations (do not know why) and without account informations I cannot connect ;-) There is a .cfg file, where all other settings have been saved to, but not the account informations. Do you know, what I have to enter into this file, so I can input it manually? Or how I could try to save the data? ... Thank you, Martin - Original Message - From: Stuart McQuade To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, January 28, 2010 11:35 AM Subject: Re: [asterisk-users] iax softphones - not reconnecting Hi, I've used ZoiPer with our Asterisk server and not had any problems. It's quite a basic-looking client but gets the job done. I believe DIAX is another option you could try. Stuart -- From: Asterisk - thinking:systems aster...@tsy.at To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thu, 28 January, 2010 10:12:11 Subject: [asterisk-users] iax softphones - not reconnecting Hi together, I try to find a softphone (freeware) solution for Windows 32, that works without problems ... Right now I use iaxcomm wich was best, of the ones I tried. But I have one problem with it. When I turn on qualify, it will not connect to the asterisk. This is also documented and normal behaveor. But if I turn of qualify, iaxcomm does not reconnect to the server, when the server got restartet or the connection got lost. So I do not know, how I can get this problem solved. Turning on qualify would get it reconnecting, but after some time it does disconnect with the note: is now unreachable. is there any other option I can activate to get asterisk or iaxcomm reconnecting? Or does anybody know any other softphone for windows, that is freeware and maybe also brandable? hope anybody can help me to get this running... Thank you very much, Martin -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linux-based hard phones?
On Thu, Jan 28, 2010 at 09:35:37AM +, Ishfaq Malik wrote: Ken D'Ambrosio wrote: Just wondering if there are any Linux-based hard phones out there -- if so, it'd be neat to see if I couldn't take advantage of the underlying OS. Thanks, -Ken Snom phones use Linux What hardware is it exactly on those phones? What CPU? How much memory? What size of NAND/flash? Which parts of that hardware are not supported in mainline kernel? (Merely having the phone run Linux is not good enough. I need to be able to modify it) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fw: OfficeSIP Softphone
On Thu, Jan 28, 2010 at 1:56 PM, Vitali Fomine supp...@officesip.comwrote: Hello, Yes, unfortunately, the sip client lib does not support udp. Best regards, Vitali Fomine Then check you are using an Asterisk patched for TCP. -- -- Adrià Vidal -- _ -- 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] Polycom Soundpoint 300IP
Hello list, anyone have a manual for the webGUI of the above phone ? Just bought a Polycom Soundpoint IP300 and on the site of Polycom I see a user manual and a administrator's manual but none of these 2 guides explain the fields in the webGUI. Trying to understand the difference between the 'server-'variables in the 'SIP'-section and the 'server'-fields in the 'Lines'-section... Anyone can point me to the manual of Polycom ??? Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inserting white noise / music / sound file intomixmonitor
Here's one possible work-around; Since you have the length of the call (from the CDR) and know the size of the Gap, you could use SOX to split the .wav file into 2 segments, then reassemble with one of the music-on-hold files segmented into it. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: Thursday, January 28, 2010 5:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Inserting white noise / music / sound file intomixmonitor A week or so ago, I explained that we need to blank our call recording when some sensitive information like credit cards where being discussed. With the lists help, I managed to find the pause/ unpause monitor commands. That works great. However (there is always a however), what that now means is that the length of the call does not match the length of the call recording, so adding stuff like this happened at 11:04 into the call now is out by the length of time of the pause :( I was wondering if it was possible to replace the voice on either leg with a sound file or something, but only in mixmonitor, as we obviously need to hear the person talking in order to take the details. Julian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linux-based hard phones?
Hi! Snom phones use Linux What hardware is it exactly on those phones? What CPU? How much memory? What size of NAND/flash? Which parts of that hardware are not supported in mainline kernel? Looks like someone out there is working on putting OpenWRT onto a snom 820. For the 3xx models check these links to learn more: INCA-IP2 Reference Design by Infineon (PDF): http://www.ip-phone-forum.de/showpost.php?p=1329580postcount=1 (contact me off-list if you want the PDF but do not want to create a login for yourself) http://www.lantiq.com/products/enterprise/ip-phone/xwaytm-inca-ip/ http://opensnom.org/index.php/Main_Page http://www.trend-verpennt.de/index.php/Snom:hardware http://www.ip-phone-forum.de/showthread.php?t=185176 http://ippf.eu/showthread.php?p=1296300 http://wiki.gpl-devices.org/wiki/Snom_370 http://www.snom.com/en/support/download/source-code/ Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iax softphones - not reconnecting
Hello list, hi Stuart, now i updated diax to 0.915a (hope this is now the newest one) The saving of the settings now works. But I am not able to register ... It seems, that the diax even does not try to register :-( Any idea or help ... has anyone an idea? Thanks, Martin - Original Message - From: Stuart McQuade To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, January 28, 2010 11:35 AM Subject: Re: [asterisk-users] iax softphones - not reconnecting Hi, I've used ZoiPer with our Asterisk server and not had any problems. It's quite a basic-looking client but gets the job done. I believe DIAX is another option you could try. Stuart -- From: Asterisk - thinking:systems aster...@tsy.at To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thu, 28 January, 2010 10:12:11 Subject: [asterisk-users] iax softphones - not reconnecting Hi together, I try to find a softphone (freeware) solution for Windows 32, that works without problems ... Right now I use iaxcomm wich was best, of the ones I tried. But I have one problem with it. When I turn on qualify, it will not connect to the asterisk. This is also documented and normal behaveor. But if I turn of qualify, iaxcomm does not reconnect to the server, when the server got restartet or the connection got lost. So I do not know, how I can get this problem solved. Turning on qualify would get it reconnecting, but after some time it does disconnect with the note: is now unreachable. is there any other option I can activate to get asterisk or iaxcomm reconnecting? Or does anybody know any other softphone for windows, that is freeware and maybe also brandable? hope anybody can help me to get this running... Thank you very much, Martin -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iax softphones - not reconnecting
okay, got it running, had to restart the application ;-) But the voice quality is very bad. :-( Does anybody know another iax client that has also a good voice quality? cheers, Martin - Original Message - From: Asterisk - thinking:systems To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, January 28, 2010 3:59 PM Subject: Re: [asterisk-users] iax softphones - not reconnecting Hello list, hi Stuart, now i updated diax to 0.915a (hope this is now the newest one) The saving of the settings now works. But I am not able to register ... It seems, that the diax even does not try to register :-( Any idea or help ... has anyone an idea? Thanks, Martin - Original Message - From: Stuart McQuade To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, January 28, 2010 11:35 AM Subject: Re: [asterisk-users] iax softphones - not reconnecting Hi, I've used ZoiPer with our Asterisk server and not had any problems. It's quite a basic-looking client but gets the job done. I believe DIAX is another option you could try. Stuart From: Asterisk - thinking:systems aster...@tsy.at To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thu, 28 January, 2010 10:12:11 Subject: [asterisk-users] iax softphones - not reconnecting Hi together, I try to find a softphone (freeware) solution for Windows 32, that works without problems ... Right now I use iaxcomm wich was best, of the ones I tried. But I have one problem with it. When I turn on qualify, it will not connect to the asterisk. This is also documented and normal behaveor. But if I turn of qualify, iaxcomm does not reconnect to the server, when the server got restartet or the connection got lost. So I do not know, how I can get this problem solved. Turning on qualify would get it reconnecting, but after some time it does disconnect with the note: is now unreachable. is there any other option I can activate to get asterisk or iaxcomm reconnecting? Or does anybody know any other softphone for windows, that is freeware and maybe also brandable? hope anybody can help me to get this running... Thank you very much, Martin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iax softphones - not reconnecting
Before blaming the client, check your settings; are you using ulaw, alaw, gsm.? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk - thinking:systems Sent: Thursday, January 28, 2010 9:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] iax softphones - not reconnecting okay, got it running, had to restart the application ;-) But the voice quality is very bad. :-( Does anybody know another iax client that has also a good voice quality? cheers, Martin - Original Message - From: Asterisk - mailto:aster...@tsy.at thinking:systems To: Asterisk Users Mailing List - mailto:asterisk-users@lists.digium.com Non-Commercial Discussion Sent: Thursday, January 28, 2010 3:59 PM Subject: Re: [asterisk-users] iax softphones - not reconnecting Hello list, hi Stuart, now i updated diax to 0.915a (hope this is now the newest one) The saving of the settings now works. But I am not able to register ... It seems, that the diax even does not try to register :-( Any idea or help ... has anyone an idea? Thanks, Martin - Original Message - From: Stuart mailto:stuart.mcqu...@yahoo.com McQuade To: Asterisk Users Mailing List - mailto:asterisk-users@lists.digium.com Non-Commercial Discussion Sent: Thursday, January 28, 2010 11:35 AM Subject: Re: [asterisk-users] iax softphones - not reconnecting Hi, I've used ZoiPer with our Asterisk server and not had any problems. It's quite a basic-looking client but gets the job done. I believe DIAX is another option you could try. Stuart _ From: Asterisk - thinking:systems aster...@tsy.at To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thu, 28 January, 2010 10:12:11 Subject: [asterisk-users] iax softphones - not reconnecting Hi together, I try to find a softphone (freeware) solution for Windows 32, that works without problems ... Right now I use iaxcomm wich was best, of the ones I tried. But I have one problem with it. When I turn on qualify, it will not connect to the asterisk. This is also documented and normal behaveor. But if I turn of qualify, iaxcomm does not reconnect to the server, when the server got restartet or the connection got lost. So I do not know, how I can get this problem solved. Turning on qualify would get it reconnecting, but after some time it does disconnect with the note: is now unreachable. is there any other option I can activate to get asterisk or iaxcomm reconnecting? Or does anybody know any other softphone for windows, that is freeware and maybe also brandable? hope anybody can help me to get this running... Thank you very much, Martin _ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] New feature in app_queue: Give members a penalty time for not answering (help testing)
Hi, I've uploaded a new patch at https://issues.asterisk.org/view.php?id=16722which adds a new option to queues.conf. The new option is notpresent-penalty, which is an amount of seconds to wait before calling a agent that doesn't answer, again. This feature is especially nice if you are using penalties on members of a queue. I would be very happy if someone could help me test this feature, and report back to the issue tracker. To test this feature, patch an asterisk-trunk source tree, set notpresent-penalty to for example 30 seconds, set up a queue with three members: penalty, member 0, member1 0, member2 1, member3 Then call into the queue, and the queue calls member1, either take the call, or neglect to answer it. He should now be unavailable for the next 30 seconds. When the other members phone rings (member2), neglect to answer this phone too. Now both agents in penalty level 0 should be unavailable, and the caller should be sent to member3. If this happends; test successfull. Håkon -- _ -- 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] rtp.c:883 ast_rtcp_read: RTCP Read too short
Hi: I have a linksys voip gateway connecting to an asterisk server ,when i dial a call from the linksys gateway to asterisk , i see repeated messages of a RTP errors ,and at same time i hear fake ring on the linksys , This is wht i see on asterisk console : -- Executing [9613070...@direct:1] Set(SIP/03070741-088bd470, CALLERID(number)=96170707070) in new stack -- Executing [9613070...@direct:2] Dial(SIP/03070741-088bd470, SIP/usa/9613070741) in new stack -- Called usa/9613070741 [Jan 28 18:17:36] WARNING[29269]: rtp.c:883 ast_rtcp_read: RTCP Read too short [Jan 28 18:17:42] WARNING[29269]: rtp.c:883 ast_rtcp_read: RTCP Read too short -- Call on SIP/usa-08906450 left from hold -- SIP/usa-08906450 is making progress passing it to SIP/03070741-088bd470 -- SIP/usa-08906450 is ringing -- Call on SIP/usa-08906450 left from hold -- SIP/usa-08906450 is making progress passing it to SIP/03070741-088bd470 [Jan 28 18:17:50] WARNING[29269]: rtp.c:883 ast_rtcp_read: RTCP Read too short [Jan 28 18:17:53] WARNING[29269]: rtp.c:883 ast_rtcp_read: RTCP Read too short [Jan 28 18:17:57] WARNING[29269]: rtp.c:883 ast_rtcp_read: RTCP Read too short -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fw: OfficeSIP Softphone
Hello, Thank you for your help. I have enable tcp by using tcpenable and tcpbindaddr. The client can not connect w/o these settings. I am trying asterisk 1.6.0.10 (in trixbox), need I install something else? Best regards, Vitali Fomine - Original Message - From: Adrià Vidal To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, January 28, 2010 4:20 PM Subject: Re: [asterisk-users] Fw: OfficeSIP Softphone On Thu, Jan 28, 2010 at 1:56 PM, Vitali Fomine supp...@officesip.com wrote: Hello, Yes, unfortunately, the sip client lib does not support udp. Best regards, Vitali Fomine Then check you are using an Asterisk patched for TCP. -- -- Adrià Vidal -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 911, location
Hi there, I am running a PBX under asterisk 1.6. I have few FXO analogue lines connecting to PSTN. These lines are in a hunt group. I trying to make my extensions to dial 91, but this is a bit scary, I mean if somebody make an emergency call after hours and without completing call is not able to tell his/her location. How can I make 911 call center to know the exact location of my extension. I think its possible by having DID's but I am looking for other options too. I would appreciate your valuable ideas and suggestions. Thanks in advance Shahnawaz -- _ -- 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] R: rtp.c:883 ast_rtcp_read: RTCP Read too short
The ring isn't fake :] The Linksys GW isn't dissing, is just responding to an INVITE. The problem is that you have problem passing voice. In other words: check RTP ports settings on server client or the firewall rules. Alex Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di wassim darwich Inviato: giovedì 28 gennaio 2010 17:38 A: asterisk-users@lists.digium.com Oggetto: [asterisk-users] rtp.c:883 ast_rtcp_read: RTCP Read too short Hi: I have a linksys voip gateway connecting to an asterisk server ,when i dial a call from the linksys gateway to asterisk , i see repeated messages of a RTP errors ,and at same time i hear fake ring on the linksys , This is wht i see on asterisk console : -- Executing [9613070...@direct:1] Set(SIP/03070741-088bd470, CALLERID(number)=96170707070) in new stack -- Executing [9613070...@direct:2] Dial(SIP/03070741-088bd470, SIP/usa/9613070741) in new stack -- Called usa/9613070741 [Jan 28 18:17:36] WARNING[29269]: rtp.c:883 ast_rtcp_read: RTCP Read too short [Jan 28 18:17:42] WARNING[29269]: rtp.c:883 ast_rtcp_read: RTCP Read too short -- Call on SIP/usa-08906450 left from hold -- SIP/usa-08906450 is making progress passing it to SIP/03070741-088bd470 -- SIP/usa-08906450 is ringing -- Call on SIP/usa-08906450 left from hold -- SIP/usa-08906450 is making progress passing it to SIP/03070741-088bd470 [Jan 28 18:17:50] WARNING[29269]: rtp.c:883 ast_rtcp_read: RTCP Read too short [Jan 28 18:17:53] WARNING[29269]: rtp.c:883 ast_rtcp_read: RTCP Read too short [Jan 28 18:17:57] WARNING[29269]: rtp.c:883 ast_rtcp_read: RTCP Read too short -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iax softphones - not reconnecting
hi! sorry, I used ulaw and alaw for all my tests. iaxcomm has perfect sound, but does not reconnect (the only problem with this tool) ephone has also great voice quality diax has bad quality with both of the codecs... (i do not understand why) It sounds like trying to mute some noices but the dsp is turned off. And thhere are ugly ticking sounds in the back, when somebody is talking. I do not want to blame any tool, only want to find one that is working great without problems in connecting, reconnecting or voice quality. hope there is a solution... cheers, Martin - Original Message - From: Danny Nicholas To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Thursday, January 28, 2010 4:30 PM Subject: Re: [asterisk-users] iax softphones - not reconnecting Before blaming the client, check your settings; are you using ulaw, alaw, gsm.? -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk - thinking:systems Sent: Thursday, January 28, 2010 9:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] iax softphones - not reconnecting okay, got it running, had to restart the application ;-) But the voice quality is very bad. :-( Does anybody know another iax client that has also a good voice quality? cheers, Martin - Original Message - From: Asterisk - thinking:systems To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, January 28, 2010 3:59 PM Subject: Re: [asterisk-users] iax softphones - not reconnecting Hello list, hi Stuart, now i updated diax to 0.915a (hope this is now the newest one) The saving of the settings now works. But I am not able to register ... It seems, that the diax even does not try to register :-( Any idea or help ... has anyone an idea? Thanks, Martin - Original Message - From: Stuart McQuade To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, January 28, 2010 11:35 AM Subject: Re: [asterisk-users] iax softphones - not reconnecting Hi, I've used ZoiPer with our Asterisk server and not had any problems. It's quite a basic-looking client but gets the job done. I believe DIAX is another option you could try. Stuart -- From: Asterisk - thinking:systems aster...@tsy.at To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thu, 28 January, 2010 10:12:11 Subject: [asterisk-users] iax softphones - not reconnecting Hi together, I try to find a softphone (freeware) solution for Windows 32, that works without problems ... Right now I use iaxcomm wich was best, of the ones I tried. But I have one problem with it. When I turn on qualify, it will not connect to the asterisk. This is also documented and normal behaveor. But if I turn of qualify, iaxcomm does not reconnect to the server, when the server got restartet or the connection got lost. So I do not know, how I can get this problem solved. Turning on qualify would get it reconnecting, but after some time it does disconnect with the note: is now unreachable. is there any other option I can activate to get asterisk or iaxcomm reconnecting? Or does anybody know any other softphone for windows, that is freeware and maybe also brandable? hope anybody can help me to get this running... Thank you very much, Martin -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- _ -- Bandwidth and Colocation Provided by
Re: [asterisk-users] Connecting to an External EPBX without an SIP provider
This all depends on your EPBX... For example 1) If you put a 2 port FXO card in an Asterisk server, you need 2 FXS ports on your EPBX to connect to 2) If you put a 4 port FXO card in an Asterisk server, you need 4 FXS ports on your EPBX to connect to 3) If you put a T1/E1 card in an Asterisk server, you need a matching T1/E1 port on your EPBX to connect to Don't hesitate to email me directly if you still have questions... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Siju George Sent: Wednesday, January 27, 2010 11:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Connecting to an External EPBX without an SIP provider Thanks for the reply jamie :-) Does ordinary EPBXs in US have those ports or do you need special EPBXs? --Siju On Wed, Jan 27, 2010 at 8:32 PM, Jamie A. Stapleton jstaple...@computer-business.com wrote: In this case, a SIP provider would not be required. Obviously, you will need ports on your EPBX to connect the Digium card to. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Siju George Sent: Wednesday, January 27, 2010 5:01 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Connecting to an External EPBX without an SIP provider Hi, If I get a Dignum Card and fit it into my computer do I still need an SIP provider to connect through my EPBX to a Public Telephone System? Thanks --Siju -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Connected to definity errors
yes G3. The problem is not getting it to work, but to have a stable working one. They work but around once a day it has to be restarted. On Wed, Jan 27, 2010 at 6:31 PM, Steve Totaro stot...@first-notification.com wrote: Definity what? G3? I did that once, a real pain but doable. I don't remember the settings but if I had a terminal in front of me, I am sure I could get it work. Thanks, Steve T On Wed, Jan 27, 2010 at 5:42 PM, C F shma...@gmail.com wrote: We didn't fix it yet. For the moment the Definity is not connected directly to Asterisk, we route all communications between Asterisk and the Definity over the PSTN. The plan is to play around with all protocol settings to figure out which one is the most stable, from what I understand - however I haven't yet tested it - att custom should work best. But we didn't yet get around to it. On Wed, Jan 27, 2010 at 12:47 PM, Alec Davis siva...@paradise.net.nz wrote: Did you get this resolved? And how if you did. We've been have the same random PRI lockup issue for years now. I've opened a mantis bug https://issues.asterisk.org/view.php?id=16713 and hopefully we can get this issue resolved. Alec -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F Sent: Thursday, 20 August 2009 11:21 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] PRI Connected to definity errors We have setup asterisk to handle our calls before between telco and an Avaya definity. The PRI keeps locking up every so often. In addition I keep getting this error when trying to call the avaya: -- Channel 0/2, span 1 got hangup request, cause 102 -- Hungup 'Zap/2-1' When that error happens I get a fast busy (congestion) tone. Any one can point me in the right direction? TIA ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911, location
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 mir shahnawaz wrote: Hi there, I am running a PBX under asterisk 1.6. I have few FXO analogue lines connecting to PSTN. These lines are in a hunt group. I trying to make my extensions to dial 91, but this is a bit scary, I mean if somebody make an emergency call after hours and without completing call is not able to tell his/her location. How can I make 911 call center to know the exact location of my extension. I think its possible by having DID's but I am looking for other options too. I would appreciate your valuable ideas and suggestions. If you're using POTS lines to make the call to 911 they'll know the location, if the POTS lines come into the building that you're calling from. Are you saying that these lines are located in a different location? Barry - -- -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFLYdtpCFu3bIiwtTARAhZyAJ941RBJz615PkJkOBLkWF8WalaMTgCfRsdR UnWTQQ1anTXtDqfk54QVj/k= =LtAE -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911, location
Thanks for your reply. Yes POTS lines are coming into the building but I have multiple rooms. Suppose a person is working late hours and have a heart attack. How could 911 locate the room when no or few people around. Thanks Smir On Thu, Jan 28, 2010 at 11:46 AM, Barry L. Kline blkl...@attglobal.net wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 mir shahnawaz wrote: Hi there, I am running a PBX under asterisk 1.6. I have few FXO analogue lines connecting to PSTN. These lines are in a hunt group. I trying to make my extensions to dial 91, but this is a bit scary, I mean if somebody make an emergency call after hours and without completing call is not able to tell his/her location. How can I make 911 call center to know the exact location of my extension. I think its possible by having DID's but I am looking for other options too. I would appreciate your valuable ideas and suggestions. If you're using POTS lines to make the call to 911 they'll know the location, if the POTS lines come into the building that you're calling from. Are you saying that these lines are located in a different location? Barry - -- -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFLYdtpCFu3bIiwtTARAhZyAJ941RBJz615PkJkOBLkWF8WalaMTgCfRsdR UnWTQQ1anTXtDqfk54QVj/k= =LtAE -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MYSQL problem
snip However, if you're going to be doing massive joins for reporting, you're better off using something else (or running individual MySQL slaves, whose purpose is to run those complex queries and doing nothing else). In a past life, our MySQL database ran circles around Oracle, Informix, and DB2... until someone ran a massive join on the same server, which caused MySQL to crawl. /snip Good distinction to make. I should have been more clear. I believe mysql has read only slave capabilities within a clustered environment, so your point about the slaves isn't out of the question. However I don't believe in database engines doing really anything other than transaction processing. That's why IMHO there should always be a distinction between the database backend and whatever software you're using to generate OLAP data (this software should NOT be the database engine). I know this is not a common opinion, but if we keep the database engine doing what it's good at and leave any report processing to external software, we're generally able to get better performance out of each individual piece... -Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911, location
Here's one solution:
- exten = _911,1,Set(IMAT=EXTEN)
- exten = _911,2,Set(IMAT=CUT(IMAT|\/|2)
- exten = _911,3,Dial(DAHDI/1,w911)
- exten = _911,4,Background(emergencyin${IMAT})
Where you would record /var/lib/asterisk/sound/emergencyin100 for extension
100, etc.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mir shahnawaz
Sent: Thursday, January 28, 2010 12:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 911, location
Thanks for your reply. Yes POTS lines are coming into the building but
I have multiple rooms. Suppose a person is working late hours and have
a heart attack. How could 911 locate the room when no or few people
around.
Thanks
Smir
On Thu, Jan 28, 2010 at 11:46 AM, Barry L. Kline blkl...@attglobal.net
wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
mir shahnawaz wrote:
Hi there,
I am running a PBX under asterisk 1.6. I have few FXO analogue lines
connecting to PSTN. These lines are in a hunt group. I trying to make
my extensions to dial 91, but this is a bit scary, I mean if somebody
make an emergency call after hours and without completing call is not
able to tell his/her location. How can I make 911 call center to know
the exact location of my extension. I think its possible by having
DID's but I am looking for other options too. I would appreciate your
valuable ideas and suggestions.
If you're using POTS lines to make the call to 911 they'll know the
location, if the POTS lines come into the building that you're calling
from. Are you saying that these lines are located in a different
location?
Barry
- --
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Version: GnuPG v1.4.5 (GNU/Linux)
iD8DBQFLYdtpCFu3bIiwtTARAhZyAJ941RBJz615PkJkOBLkWF8WalaMTgCfRsdR
UnWTQQ1anTXtDqfk54QVj/k=
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Re: [asterisk-users] 911, location
You should phone up the emergency people on a non-emergency number and ask them about that as well. On Thu, Jan 28, 2010 at 10:58 AM, mir shahnawaz shahnawaz...@gmail.com wrote: Thanks for your reply. Yes POTS lines are coming into the building but I have multiple rooms. Suppose a person is working late hours and have a heart attack. How could 911 locate the room when no or few people around. Thanks Smir On Thu, Jan 28, 2010 at 11:46 AM, Barry L. Kline blkl...@attglobal.net wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 mir shahnawaz wrote: Hi there, I am running a PBX under asterisk 1.6. I have few FXO analogue lines connecting to PSTN. These lines are in a hunt group. I trying to make my extensions to dial 91, but this is a bit scary, I mean if somebody make an emergency call after hours and without completing call is not able to tell his/her location. How can I make 911 call center to know the exact location of my extension. I think its possible by having DID's but I am looking for other options too. I would appreciate your valuable ideas and suggestions. If you're using POTS lines to make the call to 911 they'll know the location, if the POTS lines come into the building that you're calling from. Are you saying that these lines are located in a different location? Barry - -- -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFLYdtpCFu3bIiwtTARAhZyAJ941RBJz615PkJkOBLkWF8WalaMTgCfRsdR UnWTQQ1anTXtDqfk54QVj/k= =LtAE -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM2400 card FXS problems
We have a recently deployed server with a new TDM2400 card that will not put dialtone or audio on FXS ports after the physical server restarts (though they will ring if called, there's just no audio on the line if the phone at the other end picks up). The symptom can be resolved by stopping Asterisk, restarting DAHDI, and then restarting Asterisk. So far, this has happened on both times the server has been restarted (once planned, once unplanned) since the system was deployed and the phone lines were punched down to the block that is connected to the TDM card. Does anyone have suggestions on where I should start trying to troubleshoot the root cause of the FXS problem? Obviously having to manually restart Asterisk/DAHDI every time the server reboots isn't a practical long term solution. Thank you, Noah Engelberth Direct Link Computer Systems -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [inter-pbx commnication] trying to make PBX1 talk to PBX2
Hi William, thank you very much for your response, actually i used the same config but i removed the mention of the context, and it went through! 2010/1/26 William Stillwell (Lists) william.stillwell-li...@ablebody.net This is how I did it.. I have to Servers, SRV1 and SRV2 In SRV1 iax.conf [SRV1-SRV2] type=peer username=SRV1-SRV2 secret=Password1 host=IP OF SRV2 qualify=yes [SRV2-SRV1] type=user username=SRV2-SRV1 secret=Password2 context=from-iax host=IP OF SRV2 quailfy=yes If I need to make calls on other box, I do Dial(IAX2/SRV1-SRV2/XX) where X is in destination “from-iax” context On SRV2 iax.conf [SRV1-SRV2] type=user username= SRV1-SRV2 secret=Password1 host=IP of SRV1 context=from-iax qualify=yes [SRV2-SRV1] type=peer username= SRV2-SRV1 secret=Password2 host=IP of SRV1 qualify=yes And calls from Here to There are Dial(IAX2/SRV2-SRV1/X) where is in destination “from-iax” context *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati *Sent:* Tuesday, January 26, 2010 10:11 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] [inter-pbx commnication] trying to make PBX1 talk to PBX2 Hi All, i want to make an extension from pbx1 able to tlak to another extension from pbx2 or use pbx2's trunk to dial outside calls. so i edited in both servers accordinally the iax.conf: .. .. .. when i type iax2 show peers i notice that pbx's are registred. of course still didn't attend my goal, do anybody have an idea how to make this happend?! -- Abdullah -- _ -- 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 -- Abdullah -- _ -- 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] Dial cellphone from one PBX1 to PBX2? is it possible?
Hi Guys, i am using two PBX's i can call from pbx1 a cellphone tied to pbx2 that way: 1) use a phone in PBX1 2) call extension in PBX2 3) extensions in PBX2 ring to a cellphone (as this specific ext is tied to a cellphone) my questions now is : am i gonna be able to dial from an IPphone registered within PBX1 to a cellphone by using the Trunk (Zap or dahdi) of PBX2? anybody know IPphone-PBX1-IAXPBX2PRI line---cellphone??? thank you for you help guys!! -- Abdullah -- _ -- 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] rtp.c:883 ast_rtcp_read: RTCP Read too short
Hi: Firewall is disabled ,so no need to worry about firewall,but i dont know where to check rtp settings and what do i need to search for ,can you guide me please. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911, location
mir shahnawaz wrote:
Thanks, Could you please explain this little bit more. I am not
getting IMAT=EXTEN.
On Thu, Jan 28, 2010 at 12:15 PM, Danny Nicholasda...@debsinc.com wrote:
Here's one solution:
- exten = _911,1,Set(IMAT=EXTEN)
He probably meant ${EXTEN}
Doug
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Re: [asterisk-users] 911, location
Thanks, Could you please explain this little bit more. I am not
getting IMAT=EXTEN.
On Thu, Jan 28, 2010 at 12:15 PM, Danny Nicholas da...@debsinc.com wrote:
Here's one solution:
- exten = _911,1,Set(IMAT=EXTEN)
- exten = _911,2,Set(IMAT=CUT(IMAT|\/|2)
- exten = _911,3,Dial(DAHDI/1,w911)
- exten = _911,4,Background(emergencyin${IMAT})
Where you would record /var/lib/asterisk/sound/emergencyin100 for extension
100, etc.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mir shahnawaz
Sent: Thursday, January 28, 2010 12:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 911, location
Thanks for your reply. Yes POTS lines are coming into the building but
I have multiple rooms. Suppose a person is working late hours and have
a heart attack. How could 911 locate the room when no or few people
around.
Thanks
Smir
On Thu, Jan 28, 2010 at 11:46 AM, Barry L. Kline blkl...@attglobal.net
wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
mir shahnawaz wrote:
Hi there,
I am running a PBX under asterisk 1.6. I have few FXO analogue lines
connecting to PSTN. These lines are in a hunt group. I trying to make
my extensions to dial 91, but this is a bit scary, I mean if somebody
make an emergency call after hours and without completing call is not
able to tell his/her location. How can I make 911 call center to know
the exact location of my extension. I think its possible by having
DID's but I am looking for other options too. I would appreciate your
valuable ideas and suggestions.
If you're using POTS lines to make the call to 911 they'll know the
location, if the POTS lines come into the building that you're calling
from. Are you saying that these lines are located in a different
location?
Barry
- --
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.5 (GNU/Linux)
iD8DBQFLYdtpCFu3bIiwtTARAhZyAJ941RBJz615PkJkOBLkWF8WalaMTgCfRsdR
UnWTQQ1anTXtDqfk54QVj/k=
=LtAE
-END PGP SIGNATURE-
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Re: [asterisk-users] TDM2400 card FXS problems
On Thu, Jan 28, 2010 at 07:25:18PM +, Noah I. Engelberth wrote: We have a recently deployed server with a new TDM2400 card that will not put dialtone or audio on FXS ports after the physical server restarts What's the output of lsdahdi in that case? (though they will ring if called, there's just no audio on the line if the phone at the other end picks up). The symptom can be resolved by stopping Asterisk, restarting DAHDI, and then restarting Asterisk. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Use of 603 Declined
Hello everyone,
I've had the time to examine some specific serial/parallel forking
scenarios with Asterisk lately. Looking at chan_sip it appears that
anytime Asterisk wants to tear down a call before it's brought up, it
sends a 603 Declined:
} else {/* Incoming call, not up */
const char *res;
if (p-hangupcause (res =
hangup_cause2sip(p-hangupcause)))
transmit_response_reliable(p,
res, p-initreq);
else
transmit_response_reliable(p,
603 Declined, p-initreq);
p-invitestate = INV_TERMINATED;
Obviously this doesn't include cases where the URI is not found, the
codec is incompatible, etc. More just general failure stuff like
executing Hangup() on an unanswered channel.
However, 6xxx responses are somewhat religious/political in the SIP
sphere... Being that they are global responses, how could this
single Asterisk instance know that this call is unacceptable
everywhere/anywhere? From RFC3261:
21.6.2 603 Decline
The callee's machine was successfully contacted but the user
explicitly does not wish to or cannot participate. The response MAY
indicate a better time to call in the Retry-After header field. This
status response is returned only if the client knows that no other
end point will answer the request.
I suppose manually executing Hangup() justifies the first statement
but it's the last sentence that bothers me:
returned only if the client (Asterisk) knows that no other end point
will answer the request
That's a little presumptive of the Asterisk system, don't you think?
;) While I don't have any better alternative responses I'm just
bothered by the global nature of 6xx failures in the first place.
Any thoughts?
--
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com
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Re: [asterisk-users] Use of 603 Declined
Kristian, Unfortunately, 603 Declined is frequently misused this way by service providers and SIP stacks. It seems to be a catch-all epithet for some sort of miscellaneous call completion failure that cannot be categorised any other way, much like 503 Service Unavailable. I agree 100% with this statement, and have taken this position for a long time: On 01/28/2010 04:17 PM, Kristian Kielhofner wrote: returned only if the client (Asterisk) knows that no other end point will answer the request That's a little presumptive of the Asterisk system, don't you think? ;) While I don't have any better alternative responses I'm just bothered by the global nature of 6xx failures in the first place. It's also problematic because a 3261-compliant SIP proxy or UAC is not going to attempt to reach the destination by alternate means (serial forking in the case of the proxy, or a new call leg in the case of the UA) because of this precise implication of 6xx-class final replies. -- Alex -- Alex Balashov - Principal Evariste Systems LLC Tel: +1 678-954-0670 Direct : +1 678-954-0671 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AsyncGoto/DAHDI ?
Usually I see /DAHDI/*channel #*, but today I see this AsyncGoto/DAHDI/*channel#* on one of my call. What does this mean? -- _ -- 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] How to set sip client idle or busy in Asterisk ?
Hello every one, I just want to add a soft button to make my soft sip client with idle or busy status. Does any one know what's the event action drive Asterisk to be busy or idle in API event list? Thanks, Johnson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Use of 603 Declined
On Thu, Jan 28, 2010 at 4:23 PM, Alex Balashov abalas...@evaristesys.com wrote: It's also problematic because a 3261-compliant SIP proxy or UAC is not going to attempt to reach the destination by alternate means (serial forking in the case of the proxy, or a new call leg in the case of the UA) because of this precise implication of 6xx-class final replies. -- Alex This is precisely why some proxies (including OpenSIPS Kamailio) have added the disable_6xx_block parameter to specifically break this 3261-compliant behavior. Of course this being a global proxy parameter prevents cases where you really do want a 603 to stop forking. I've read that OpenSIPS is going to make it possible to activate this behavior via flags or some other means but in the meantime I'd like to see Asterisk be a little more flexible and um, friendly in this case. Luckily Asterisk is open source and I can make that change if I like but... A quick poll: Who thinks Asterisk should severely limit the cases it sends 6xx responses? -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Use of 603 Declined
On 01/28/2010 04:47 PM, Kristian Kielhofner wrote: On Thu, Jan 28, 2010 at 4:23 PM, Alex Balashov abalas...@evaristesys.com wrote: It's also problematic because a 3261-compliant SIP proxy or UAC is not going to attempt to reach the destination by alternate means (serial forking in the case of the proxy, or a new call leg in the case of the UA) because of this precise implication of 6xx-class final replies. -- Alex This is precisely why some proxies (including OpenSIPS Kamailio) have added the disable_6xx_block parameter to specifically break this 3261-compliant behavior. Of course this being a global proxy parameter prevents cases where you really do want a 603 to stop forking. I've read that OpenSIPS is going to make it possible to activate this behavior via flags or some other means but in the meantime I'd like to see Asterisk be a little more flexible and um, friendly in this case. Luckily Asterisk is open source and I can make that change if I like but... I was just about to mention the disable_6xx_block parameter, but figured it would be too pedantic/off-topic for this thread. A quick poll: Who thinks Asterisk should severely limit the cases it sends 6xx responses? I can't think of any cases where it should be used except where some sort of formal error arises, to be honest. When is Asterisk ever in an authoritative position to deem a destination certifiably unreachable except, perhaps, an invalid IP address, unresolvable host, or something of that sort? -- Alex Balashov - Principal Evariste Systems LLC Tel: +1 678-954-0670 Direct : +1 678-954-0671 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial cellphone from one PBX1 to PBX2? is it possible?
Your inbound context needs to have access to your outbound context.
[iax-inbound]
Include = outbound-conext
[outbound-context]
Exten = _1NXXNXX,1,Dial(DAHDI\g1\${EXTEN})
Something like that.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati
Sent: Thursday, January 28, 2010 3:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Dial cellphone from one PBX1 to PBX2? is it
possible?
Hi Guys,
i am using two PBX's i can call from pbx1 a cellphone tied to pbx2 that way:
1) use a phone in PBX1
2) call extension in PBX2
3) extensions in PBX2 ring to a cellphone (as this specific ext is tied to a
cellphone)
my questions now is : am i gonna be able to dial from an IPphone registered
within PBX1 to a cellphone by using the Trunk (Zap or dahdi) of PBX2?
anybody know
IPphone-PBX1-IAXPBX2PRI
line---cellphone???
thank you for you help guys!!
--
Abdullah
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Re: [asterisk-users] Use of 603 Declined
On Thu, Jan 28, 2010 at 4:52 PM, Alex Balashov abalas...@evaristesys.com wrote: I was just about to mention the disable_6xx_block parameter, but figured it would be too pedantic/off-topic for this thread. I didn't. Google has a great memory and hopefully now when some poor soul is researching this (Asterisk + OpenSIPS/Kamailio 6xx replies) they will find this thread to tie their solution together. I can't think of any cases where it should be used except where some sort of formal error arises, to be honest. When is Asterisk ever in an authoritative position to deem a destination certifiably unreachable except, perhaps, an invalid IP address, unresolvable host, or something of that sort? Agreed. Even then, on an incoming request, how would it know? -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iax client for symbian s60
On 28/01/10 9:14 PM, Asterisk - thinking:systems wrote: Hi all, I searched for a long time and know that here this question also was asked in the past, but ... Is there any iax client for s60 now? Or still no client available? There are so many people asking for it, but nobody seems to get it done :-( Not that I'm aware of - best place to ask would be the IAXClient mailing list, but I'm pretty sure I'd remember if someone had written one. Probably the closest would be Tim Panton's work - maybe hunt him down :D -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- 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] Cell Phone dialing
Greetings all, This was most likely covered in one or more of the 15K emails I tried to categorize today. I'm running * 1.4.26.2 with TDM400P. When I call number 205-555-1212 (a land line), Asterisk indicates ringing after about 2-3 seconds. When I call 205-555-1313 (a cell phone), it takes 4-5 seconds to indicate. Is this a known problem and/or something I have to live with? Regards, Danny Nicholas -- _ -- 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] beroNet BN4S0e PCI Express ISDN Card with chan_dahdi
Hi, I'm currently trying to get a BN4S0e (which is basically a BN4S0 with a PCIe connector) working with dahdi. The module is loading properly but the card is not detected by the module. Is support on dahdi planned for this card ? In the meantime i'm gonna use mISDN with this card. Thanks Laurent -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cell Phone dialing
What happens when you dial with a handset? Is this delay caused by the asterisk or is the telco doing it? On Thu, Jan 28, 2010 at 2:57 PM, Danny Nicholas da...@debsinc.com wrote: Greetings all, This was most likely covered in one or more of the 15K emails I tried to categorize today. I’m running * 1.4.26.2 with TDM400P. When I call number 205-555-1212 (a land line), Asterisk indicates ringing after about 2-3 seconds. When I call 205-555-1313 (a cell phone), it takes 4-5 seconds to indicate. Is this a known problem and/or something I have to live with? Regards, Danny Nicholas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unable to create channel of type 'DAHDI' (cause 0 - Unknown)
Hi, I have a tdm22b (2 fxs / 2 fxo) When Asterisk is just started, outbound calls routing to fxo port, do not working with error: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) Inbound calls to fxo port work fine. After first inbound call, the outbound calls starts working. CentOS 5.4 asterisk 1.6.0.21-1 dahdi 2.2.1.-1 Can anybody help me to identify what is the possible cause of problem? Thanks, Walter. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to create channel of type 'DAHDI' (cause 0- Unknown)
I have had the exact same problem for over a year on my server sporting the TDM800 but NOT on my server with the TE212P. Both servers run the same version of Linux, Asterisk and DHADI. The problem has remained consistent through all versions of DAHDI 2.0.x through 2.2.0.2, and every version of Asterisk which I have I've tried which includes various iterations of 1.6.0, 1.6.1, and 1.6.2. Currently 1.2.6.1. Surprisingly I also observe that I can even compile install NEW versions of Asterisk and/or DAHDI, and NOT observe the bug provided that I do NOT bounce the server. A developer (not an asterisk developer) named Jim Duda posted this issue to the list back in October of 08. (Asterisk 1.6.0-beta9 DAHDI 2.0.x originally for him). After what he described as considerable effort he found that by changing one line in chan_dahdi.c the issue appeared to be resolved (below). His simple patch (below) has works (for me too) as a stop-gap. I posted this the DEV list back January of 09, and the issue was reopened and then closed as 'fixed' . It would appear the issue needs to be re-reopened, as it's now appearing less specific to my hardware or configuration. https://issues.asterisk.org/view.php?id=13786 Duplicate issue to 13927 If I do not patch chan_dahdi (below), this is what I (still) observe: ONLY after a system reboot, any attempt(s) to dial from a device attached to an FXS port on my TDM800P, result in the following error : WARNING[2975]: app_dial.c:1502 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) BUT after the first INBOUND call to any FXO port on the device, the FXS port works normally until the next reboot. (Asterisk 1.6.2.1 DAHDI 2.2.0.2 ( earlier ) Centos 2.6.18-164.11.1.el5 #1 SMP Wed Jan 20 07:39:04 EST 2010 i686 i686 i386 GNU/Linux Does anyone else observe this? Could it be specific to certain (mis)configurations? It's possible that others have the issue but do not know it. With any inbound call volume it may be nearly transparent :-) -Karl JIM's ONE-LINE FIX On line 8730 (I think it's still on this line) of chan_dahdi.c replace a return 0 with return 1. if (par.rxisoffhook) return 1; else - return 0; + return 1; - Original Message - From: Walter Arguello walter_argue...@yahoo.com To: asterisk-users@lists.digium.com Sent: Thursday, January 28, 2010 6:15 PM Subject: [asterisk-users] Unable to create channel of type 'DAHDI' (cause 0- Unknown) Hi, I have a tdm22b (2 fxs / 2 fxo) When Asterisk is just started, outbound calls routing to fxo port, do not working with error: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) Inbound calls to fxo port work fine. After first inbound call, the outbound calls starts working. CentOS 5.4 asterisk 1.6.0.21-1 dahdi 2.2.1.-1 Can anybody help me to identify what is the possible cause of problem? Thanks, Walter. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to create channel of type 'DAHDI' (cause 0- Unknown)
Interesting. I had the same problem last Sept with a TDM800, DAHDI 2.2.0.2. Shaun Ruffel of Digium pointed me to https://issues.asterisk.org/file_download.php?file_id=22725type=bug which fixed it for me. This fix is already in 2.2.1. -- Barry On Thu, Jan 28, 2010 at 07:30:57PM -0600, Karl Fife wrote: I have had the exact same problem for over a year on my server sporting the TDM800 but NOT on my server with the TE212P. Both servers run the same version of Linux, Asterisk and DHADI. The problem has remained consistent through all versions of DAHDI 2.0.x through 2.2.0.2, and every version of Asterisk which I have I've tried which includes various iterations of 1.6.0, 1.6.1, and 1.6.2. Currently 1.2.6.1. Surprisingly I also observe that I can even compile install NEW versions of Asterisk and/or DAHDI, and NOT observe the bug provided that I do NOT bounce the server. A developer (not an asterisk developer) named Jim Duda posted this issue to the list back in October of 08. (Asterisk 1.6.0-beta9 DAHDI 2.0.x originally for him). After what he described as considerable effort he found that by changing one line in chan_dahdi.c the issue appeared to be resolved (below). His simple patch (below) has works (for me too) as a stop-gap. I posted this the DEV list back January of 09, and the issue was reopened and then closed as 'fixed' . It would appear the issue needs to be re-reopened, as it's now appearing less specific to my hardware or configuration. https://issues.asterisk.org/view.php?id=13786 Duplicate issue to 13927 If I do not patch chan_dahdi (below), this is what I (still) observe: ONLY after a system reboot, any attempt(s) to dial from a device attached to an FXS port on my TDM800P, result in the following error : WARNING[2975]: app_dial.c:1502 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) BUT after the first INBOUND call to any FXO port on the device, the FXS port works normally until the next reboot. (Asterisk 1.6.2.1 DAHDI 2.2.0.2 ( earlier ) Centos 2.6.18-164.11.1.el5 #1 SMP Wed Jan 20 07:39:04 EST 2010 i686 i686 i386 GNU/Linux Does anyone else observe this? Could it be specific to certain (mis)configurations? It's possible that others have the issue but do not know it. With any inbound call volume it may be nearly transparent :-) -Karl JIM's ONE-LINE FIX On line 8730 (I think it's still on this line) of chan_dahdi.c replace a return 0 with return 1. if (par.rxisoffhook) return 1; else - return 0; + return 1; - Original Message - From: Walter Arguello walter_argue...@yahoo.com To: asterisk-users@lists.digium.com Sent: Thursday, January 28, 2010 6:15 PM Subject: [asterisk-users] Unable to create channel of type 'DAHDI' (cause 0- Unknown) Hi, I have a tdm22b (2 fxs / 2 fxo) When Asterisk is just started, outbound calls routing to fxo port, do not working with error: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) Inbound calls to fxo port work fine. After first inbound call, the outbound calls starts working. CentOS 5.4 asterisk 1.6.0.21-1 dahdi 2.2.1.-1 Can anybody help me to identify what is the possible cause of problem? Thanks, Walter. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911, location
- Doug Lytle supp...@drdos.info wrote:
mir shahnawaz wrote:
Thanks, Could you please explain this little bit more. I am not
getting IMAT=EXTEN.
On Thu, Jan 28, 2010 at 12:15 PM, Danny Nicholasda...@debsinc.com
wrote:
Here's one solution:
- exten = _911,1,Set(IMAT=EXTEN)
He probably meant ${EXTEN}
Doug
If nobody is around how would they even get into the building ? Certainly in
the UK nobody should ever be in the building on their own for this exact
reason; and if they are then in would be prudent to have man down alarms and
paging.
--
Thanks, Phil
--
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Re: [asterisk-users] Unable to create channel of type 'DAHDI'(cause 0- Unknown)
Appears completely resolved! No more home-spun patches! Thanks! -K - Original Message - From: Barry Miller asterisk-us...@notanet.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, January 28, 2010 10:40 PM Subject: Re: [asterisk-users] Unable to create channel of type 'DAHDI'(cause 0- Unknown) Interesting. I had the same problem last Sept with a TDM800, DAHDI 2.2.0.2. Shaun Ruffel of Digium pointed me to https://issues.asterisk.org/file_download.php?file_id=22725type=bug which fixed it for me. This fix is already in 2.2.1. -- Barry On Thu, Jan 28, 2010 at 07:30:57PM -0600, Karl Fife wrote: I have had the exact same problem for over a year on my server sporting the TDM800 but NOT on my server with the TE212P. Both servers run the same version of Linux, Asterisk and DHADI. The problem has remained consistent through all versions of DAHDI 2.0.x through 2.2.0.2, and every version of Asterisk which I have I've tried which includes various iterations of 1.6.0, 1.6.1, and 1.6.2. Currently 1.2.6.1. Surprisingly I also observe that I can even compile install NEW versions of Asterisk and/or DAHDI, and NOT observe the bug provided that I do NOT bounce the server. A developer (not an asterisk developer) named Jim Duda posted this issue to the list back in October of 08. (Asterisk 1.6.0-beta9 DAHDI 2.0.x originally for him). After what he described as considerable effort he found that by changing one line in chan_dahdi.c the issue appeared to be resolved (below). His simple patch (below) has works (for me too) as a stop-gap. I posted this the DEV list back January of 09, and the issue was reopened and then closed as 'fixed' . It would appear the issue needs to be re-reopened, as it's now appearing less specific to my hardware or configuration. https://issues.asterisk.org/view.php?id=13786 Duplicate issue to 13927 If I do not patch chan_dahdi (below), this is what I (still) observe: ONLY after a system reboot, any attempt(s) to dial from a device attached to an FXS port on my TDM800P, result in the following error : WARNING[2975]: app_dial.c:1502 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) BUT after the first INBOUND call to any FXO port on the device, the FXS port works normally until the next reboot. (Asterisk 1.6.2.1 DAHDI 2.2.0.2 ( earlier ) Centos 2.6.18-164.11.1.el5 #1 SMP Wed Jan 20 07:39:04 EST 2010 i686 i686 i386 GNU/Linux Does anyone else observe this? Could it be specific to certain (mis)configurations? It's possible that others have the issue but do not know it. With any inbound call volume it may be nearly transparent :-) -Karl JIM's ONE-LINE FIX On line 8730 (I think it's still on this line) of chan_dahdi.c replace a return 0 with return 1. if (par.rxisoffhook) return 1; else - return 0; + return 1; - Original Message - From: Walter Arguello walter_argue...@yahoo.com To: asterisk-users@lists.digium.com Sent: Thursday, January 28, 2010 6:15 PM Subject: [asterisk-users] Unable to create channel of type 'DAHDI' (cause 0- Unknown) Hi, I have a tdm22b (2 fxs / 2 fxo) When Asterisk is just started, outbound calls routing to fxo port, do not working with error: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) Inbound calls to fxo port work fine. After first inbound call, the outbound calls starts working. CentOS 5.4 asterisk 1.6.0.21-1 dahdi 2.2.1.-1 Can anybody help me to identify what is the possible cause of problem? Thanks, Walter. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] R: rtp.c:883 ast_rtcp_read: RTCP Read too short
Hello Wassim, server side you can check the RTP ports configured in rtp.conf which you will find in /etc/asterisk/. If the file isn't there, here are the defaults: ;[general] ; ; RTP start and RTP end configure start and end addresses ; ; Defaults are rtpstart=5000 and rtpend=31000 You can even debug the RTP : CLI rtp debug ip xxx.xxx.xxx.xxx(linksys) Asterisk listens on one of those ports(rtp.conf ones) when a call is initiated. The same does your Linksys GW: it will listen only on the RTP configured ports. Check the firewall between the VoIP server and the Linsys GW and check the firewall on the Asterisk server. Debugging SIP you can see which ports are involved. There might be other problems, maybe because you are trying to directly pass the call from one peer(let's say an external voice provider) to the other(linksys). In that case careinvite=no is be your friend. Regards, Alex Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di wassim darwich Inviato: giovedì 28 gennaio 2010 21:41 A: asterisk-users@lists.digium.com Oggetto: [asterisk-users] rtp.c:883 ast_rtcp_read: RTCP Read too short Hi: Firewall is disabled ,so no need to worry about firewall,but i dont know where to check rtp settings and what do i need to search for ,can you guide me please. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to create channel of type 'DAHDI'(cause 0- Unknown)
On Thu, 2010-01-28 at 23:11 -0600, Karl Fife wrote: Appears completely resolved! No more home-spun patches! Thanks! -K It's *not* fixed here: DAHDI Version: 2.2.1 Echo Canceller: MG2 But as is depressingly the 'norm' for Asterisk it comes back to bitching about hardware 'buy an expensive Digium echo machine instead of a cheap one' rather than the fact that the core of Asterisk is rotten, buggy and the fix usually comes in the form of a developer arguing that it's somebody else's issue. Really - if Asterisk is 'The future of telephony' I can only assume that statement comes from the late 1800's. If you like echo, flaky connections, intermittent service and partially working DTMF coupled with a hefty hardware price tag then hey ho - Asterisk is your man Nice try, be great when it's finished. -- _ -- 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
