Re: [asterisk-users] click2call for conferencing two mobile numbers
Thanks Stiles Trying as u asked to do Regards On Fri, May 6, 2016 at 6:10 PM, A J Stiles <asterisk_l...@earthshod.co.uk> wrote: > On Friday 06 May 2016, Alok Srivastava wrote: > > Dear List > > wanna configure click2call in such a manner that my asterisk box call two > > mobile numbers and connect both numbers to talk. I have configured voip > > gateway, my internal and external calls are working fine. > > please help , > > You ought to be able to do this just using call files. > > All you have to do is inject a callfile (format is explained on the Wiki) > into the folder /var/spool/asterisk/outgoing/ . You have to do this > within a > CGI script, so you can pass the two end numbers to that script when the > button > is clicked. > > Note that depending on the block size used on the underlying device, you > probably should first create the file in some temporary location and then > mv it > to ...outgoing/ . Otherwise there is a danger of Asterisk reading an > incomplete file and doing nothing. Only if you know the entire file is > definitely going to be smaller than one block, can you get away with > creating > it in place. > > -- > AJS > > Note: Originating address only accepts e-mail from list! If replying off- > list, change address to asterisk1list at earthshod dot co dot uk . > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] click2call for conferencing two mobile numbers
Dear List wanna configure click2call in such a manner that my asterisk box call two mobile numbers and connect both numbers to talk. I have configured voip gateway, my internal and external calls are working fine. please help , abhi -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] NOT able to call on local extensions while successfully call on external mobile no.(using SONETEL account)
*Dear List*
Plz help, i am not much experienced with asterisk. i configured it on
ubuntu 12.04. no problem when i call any mobile no(0091XX) but when
i call on my local asterisk no.(101,102 or 105) it is not connecting
giving error
Auto fallthrough, channel 'SIP/lucknow-006f' status is 'CHANUNAVAIL'
*while when i call 200 it is playing audiofile successfully. Please help *here
is my sip.conf and extensions.conf.
thanks.
*=sip.conf*
[general]
context=unauthenticated
allowguest=yes
srvlookup=yes
udpbindaddr=0.0.0.0
tcpenable=no
register = supp...@mydomain.net:passw...@sip.sonetel.com
outboundproxy=sip.sonetel.com
[usa_number]
type=friend
dtmfmode=rfc2833
context=hello123
host=sip.sonetel.com
username=support
secret=password
nat=yes
fromdomain=mydomain.net
outboundproxy=sip.sonetel.com
insecure=invite
disallow=All
allow=alaw
allow=ulaw
allow=gsm
[office-phone](!)
type=friend
context=LocalSets
host=dynamic
nat=yes
secret=s3CuR#p@s5
dtmfmode=auto
disallow=all
allow=ulaw
; define a device name and use the office-phone template
[bombay](office-phone)
; define another device name using the same template
[lucknow](office-phone)
[test5](office-phone)
===
*extensions.conf===*
[LocalSets]
exten = _00X.,1, Answer
exten = _00X.,n, Set(CALLERID(num)=support)
exten = _00X.,n, Dial(SIP/${EXTEN}@support)
exten = _00X.,n, Hangup
exten = 101,1,Dial(SIP/lucknow)
exten = 102,1,Dial(SIP/bombay)
exten = 105,1,Dial(SIP/test5)
exten = 200,1,Answer()
same = n,Playback(an_audio_file)
same = n,Hangup()
[hello123]
exten = support,1,Answer
exten = support,n,Playback(Enterprise-Welcome-message);
exten = support,n, Hangup
===
regards
abhi
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Re: [asterisk-users] How to use phpmyadmin to remotely access asterisk mysql database?
yes u can access form same phpmyadmin both database, depends, for which database u entered userid and password on phpmyadmin login page. On Thu, Sep 11, 2014 at 2:06 PM, rafa alfurqan rafa.alfur...@gmail.com wrote: Hi, thank you for your repplied, As you're on Ubuntu, you can begin with $ sudo apt-get install phpmyadmin i did that, so what i have to do for the configuration in asterisk so i could remote to asterisk database from phpmyadmin? Also, 10.04 is a really old Ubuntu release now, even although it is a Long Term Support one. Consider upgrading to 14.04. You can apt-get dist-upgrade straight from an LTS release to the next LTS release, without needing to go through all the intermediate releases. really appreciate for the advice, i'll do that after i could remote to asterisk database from phpmyadmin. actually i have installed freeradius-server on my ubuntu too, and i could remote the database freeradius from phpmyadmin. is it possible if same phpmyadmin could remote database from freeradius-server and asterisk (they are on same server)? thank you -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] res_calendar.so and res_calendar_caldav.so
dear lists trying to integrate google calendar with asterisk 1.8.20.1. but 'calendar show calendars' not showing anything. when i run 'show modules' on asterisk prompt. it is not showing res_calendar_caldav.so module, only showing res_calendar.so module . is there anything wrong with google API. plz help. abhi -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] testing asterisk11 on single machine
can i test my asterisk11 on a single machine on which asterisk is installed that sounds are working from both end properly. i have installed asterisk 11 on ubuntu12.04 with twinkle soft phone. regards abhi -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] click to call
dear is there any study material for implementing click to call in asterisk. plz help. thanks regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip set debug on always showing error
dear please Help. I am continously getting this message after sip set debug on. and not getting clear voice from both side. --- Transmitting (NAT) to 122.163.193.94:1893 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.106:5060 ;branch=z9hG4bK-323331-c72a5bb7c840f6a183902fdcca79241b;received=122.163.193.94;rport=1893 From: 2002 sip:2002@122.160.154.189;tag=5a1cc54c To: 2002 sip:2002@122.160.154.189;tag=as64f1f102 Call-ID: 8c18bd84585128a3f0885f54dfa966ba@0:0:0:0:0:0:0:0 CSeq: 245 OPTIONS Server: Asterisk PBX 10.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 Scheduling destruction of SIP dialog '8c18bd84585128a3f0885f54dfa966ba@0:0:0:0:0:0:0:0' in 32000 ms (Method: OPTIONS) Really destroying SIP dialog '6636e2ee56cce3b0e2287ebe51962c84@0:0:0:0:0:0:0:0' Method: OPTIONS Really destroying SIP dialog '4ec2032c2b9a58e1cf2d1afa70c1970b@0:0:0:0:0:0:0:0' Method: OPTIONS -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip set debug on always showing error
previously i was using for codec allow=all after that i changed disallow=all allow=silk24 and i also change softph x-lite from jitsi(because of codec) now voice was coming fine from both side. But when i came to home from office not getting voice from both side. Threr is Airtel Broadband at my place. On Thu, Jul 5, 2012 at 3:36 PM, SamyGo govoi...@gmail.com wrote: Hi, *CSeq: 245 OPTIONS * * * This is just SIP keep-alive. It has nothing to do with any Call-media degradation. If you are not getting clear voice check the codecs, network latency/delay/loss/jitter parameters. BR Sammy On Thu, Jul 5, 2012 at 2:34 PM, alok srivastava alok...@gmail.com wrote: dear please Help. I am continously getting this message after sip set debug on. and not getting clear voice from both side. --- Transmitting (NAT) to 122.163.193.94:1893 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.106:5060 ;branch=z9hG4bK-323331-c72a5bb7c840f6a183902fdcca79241b;received=122.163.193.94;rport=1893 From: 2002 sip:2002@122.160.154.189;tag=5a1cc54c To: 2002 sip:2002@122.160.154.189;tag=as64f1f102 Call-ID: 8c18bd84585128a3f0885f54dfa966ba@0:0:0:0:0:0:0:0 CSeq: 245 OPTIONS Server: Asterisk PBX 10.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 Scheduling destruction of SIP dialog '8c18bd84585128a3f0885f54dfa966ba@0:0:0:0:0:0:0:0' in 32000 ms (Method: OPTIONS) Really destroying SIP dialog '6636e2ee56cce3b0e2287ebe51962c84@0:0:0:0:0:0:0:0' Method: OPTIONS Really destroying SIP dialog '4ec2032c2b9a58e1cf2d1afa70c1970b@0:0:0:0:0:0:0:0' Method: OPTIONS -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] port 5060 is blocked by ISP
thanks Samy i have set nat=yes, now getting sound from both side but there is too uch disturbance. soetime we becoe audible and sometime not.i did not set extern ip coz my asterisk server is directly configured on public ip. I have softphones on some where localnets separate from asterisk server campus . i also set sip set debug on CLI prompt. this is giving following error. when i test sip traffic on wireshark 401 unauthorize error getting this error cli prompt also showing. my first softph(9001) is on localnet 192.168.1.136 and 2nd softphone (9000) in another localnet in another campus(192.168.6.25) Scheduling destruction of SIP dialog ' 551b9e744a6b41ee2c00033e22b333c0@122.160.154.189:5060' in 32000 ms (Method: INVITE) [Jul 4 19:37:39] WARNING[3054]: chan_sip.c:3660 retrans_pkt: Retransmission timeout reached on transmission 551b9e744a6b41ee2c00033e22b333c0@122.160.154.189:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response [Jul 4 19:37:39] WARNING[3054]: chan_sip.c:3689 retrans_pkt: Hanging up call 551b9e744a6b41ee2c00033e22b333c0@122.160.154.189:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). -- SIP/9000-0005 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/9001-0004' status is 'CONGESTION' --- Reliably Transmitting (NAT) to 122.163.193.94:1801 --- SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 192.168.1.136:5060 ;branch=z9hG4bK-d8754z-db50ccbe7eb74a0a-1---d8754z-;received=122.163.193.94;rport=1801 From: 9001sip:9001@122.160.154.189;tag=b0785362 To: sip:9000@122.160.154.189;tag=as6c7d28d1 Call-ID: MGViNzg2YjNiOGQ4YzRmNDgxYmU2OTc2NDk4YjA1NGU. CSeq: 2 INVITE Server: Asterisk PBX 10.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-Asterisk-HangupCause: No user responding X-Asterisk-HangupCauseCode: 18 Content-Length: 0 Really destroying SIP dialog ' 551b9e744a6b41ee2c00033e22b333c0@122.160.154.189:5060' Method: INVITE --- SIP read from UDP:122.163.193.94:1801 --- ACK sip:9000@122.160.154.189 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.136:5060 ;branch=z9hG4bK-d8754z-db50ccbe7eb74a0a-1---d8754z-;rport Max-Forwards: 70 To: sip:9000@122.160.154.189;tag=as6c7d28d1 From: 9001sip:9001@122.160.154.189;tag=b0785362 Call-ID: MGViNzg2YjNiOGQ4YzRmNDgxYmU2OTc2NDk4YjA1NGU. CSeq: 2 ACK Content-Length: 0 - --- (8 headers 0 lines) --- Really destroying SIP dialog 'MGViNzg2YjNiOGQ4YzRmNDgxYmU2OTc2NDk4YjA1NGU.' Method: ACK --- SIP read from UDP:122.163.193.94:1801 --- - --- SIP read from UDP:115.249.67.250:5060 --- REGISTER sip:122.160.154.189 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.25;rport;branch=z9hG4bKskzxkdlp Max-Forwards: 70 To: shekhar sip:9000@122.160.154.189 From: shekhar sip:9000@122.160.154.189;tag=jcysf Call-ID: qajoimvsihxpsff@alok-Inspiron-N5110 CSeq: 954 REGISTER Contact: sip:9000@192.168.6.25;expires=3600 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE User-Agent: Twinkle/1.4.2 Content-Length: 0 - --- (11 headers 0 lines) --- Sending to 115.249.67.250:5060 (NAT) --- Transmitting (NAT) to 115.249.67.250:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.6.25;branch=z9hG4bKskzxkdlp;received=115.249.67.250;rport=5060 From: shekhar sip:9000@122.160.154.189;tag=jcysf To: shekhar sip:9000@122.160.154.189;tag=as26d4cd86 Call-ID: qajoimvsihxpsff@alok-Inspiron-N5110 CSeq: 954 REGISTER Server: Asterisk PBX 10.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=278a3764 Content-Length: 0 Scheduling destruction of SIP dialog 'qajoimvsihxpsff@alok-Inspiron-N5110' in 32000 ms (Method: REGISTER) --- SIP read from UDP:115.249.67.250:5060 --- REGISTER sip:122.160.154.189 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.25;rport;branch=z9hG4bKjmqlllxn Max-Forwards: 70 To: shekhar sip:9000@122.160.154.189 From: shekhar sip:9000@122.160.154.189;tag=jcysf Call-ID: qajoimvsihxpsff@alok-Inspiron-N5110 CSeq: 955 REGISTER Contact: sip:9000@192.168.6.25;expires=3600 Authorization: Digest username=9000,realm=asterisk,nonce=278a3764,uri=sip:122.160.154.189,response=c7a119185514202d5f9cc10a86a93607,algorithm=MD5 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE User-Agent: Twinkle/1.4.2 Content-Length: 0 - --- (12 headers 0 lines) --- Sending to 115.249.67.250:5060 (NAT) --- Transmitting (NAT) to 115.249.67.250:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.6.25;branch=z9hG4bKjmqlllxn;received=115.249.67.250;rport=5060 From: shekhar sip:9000@122.160.154.189;tag=jcysf To: shekhar sip:9000@122.160.154.189;tag=as26d4cd86 Call-ID: qajoimvsihxpsff@alok-Inspiron-N5110 CSeq: 955
[asterisk-users] port 5060 is blocked by ISP
dear i have configured properly asterisk. At the one end i am using x-lite soft ph and another end twinkle. call is going properly from both end but after picking the phone not able to listen other one. when i checked the port 5060 on the asterisk server it is always showing closed while i have flushed all the rules from iptables (iptables -F) PORT STATE SERVICE VERSION 5060/tcp closed sip telnet localhost 5060 (could not connect) regards alok -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP keepalive doesn't work
Kevin P. Fleming kpfleming at digium.com writes: Yes, it was lost during a merge of code into Asterisk trunk after 1.6.2 was branched (so only 1.8.0 and trunk are missing the code). Leif Madsen entered an issue on Mantis as a blocker for any more 1.8.x releases until this is resolved, as it is clearly a regression in the 1.8.x series. So is this working in which release 1.6.2.18 ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Showing 404 not found when calling from third party SIP server (newbie question)
Hi All, I have installed Asterisk Successfully and configure a out bound trunk for another SIP server so that if Ill dial 777123 from an asterisk-registered-phone then it will dial to the phone extension(123)-registered in the third party server. But my problem is that the reverse is not happening, that is I am not able to call from Third party SIP server to Asterisk extensions. Actually the third party SIP Server is sending request to Asterisk for the extension registered but Asterisk sending the response 404 not found where as the actually the extension really exist. I have included the all extensions contexts with the incoming contexts. Is there anything to configure for incoming contexts. Is there any way to know which trunk is accepting the incoming calls Thanks and Regards Alok Mohapatra ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk web interface is not parsing the PHP pages
Hi All, I have installed Asterisk 1.2.10 on Fedora 5. I have installed Asterisk Management Portal (AMP) for web interface. After installing properly when opening in the webpage it is not parsing the index.php for the AMP. My Database is MySQL.and web server is Apache 2.2. Please let me know is this configuration problem or this is the problem with Apache (Apache 2.2) . Thanks and Regards Alok Mohapatra ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Configuring 2 Asterisk servers with a SIP trunk
Hi All, Please let me know the how to configure a SIP trunk of a asterisk Server with another one (not IAX2). Asterisk-A should register a SIP trunk with Asterisk-B server . With Regards Alok Ranjan Mohapatra Software Engineer +91 9866269992 PrimeSoft IP Solutions (P) Ltd # 917- 922,East Wing, 9th floor Block III, White House,Begumpet Hyderabad - 500016, INDIA Ph - 91-40-23418239/40 www.primesoftindia.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sprint PCS - Asterisk through Digium TDM400P
Does anyone have trouble with dialing in to an Asterisk Server and having the DTMF digits recognized? We have some clients who are calling in with cell phones, notably those with SprintPCS service, who's DTMF is just never recognized. I have tried relax_dtmf on and off, with no improvement. My rxgain is currently set to 3. Can anyone suggest possible solutions? Incoming calls are coming through POTS lines connected to the server to TDM400P with FXO modules. Thanks, Al -- Alok K. Dhir [EMAIL PROTECTED] Symplicity Corporation http://solutions.symplicity.com 703 351 6987 (w) | 703 351-6357 (f) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] latest CVS build won't load
Fails on loading several of the chan_*.so modules with undefined symbol __use_ast_pthread_create_instead__. Notably, these same modules complain during compilation implicit declaration of function __use_ast_pthread_create_instead__. Ideas? -- Alok K. Dhir [EMAIL PROTECTED] Symplicity Corporation http://solutions.symplicity.com 703 351 6987 (w) | 703 351-6357 (f) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] latest CVS build won't load
I did... Chris Shaw wrote: Make sure you delete your /usr/lib/asterisk directory before installing a new CVS copy... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alok K. Dhir [EMAIL PROTECTED] Symplicity Corporation http://solutions.symplicity.com 703 351 6987 (w) | 703 351-6357 (f) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P lockups (FXO)
Hey all - we have two TDM400P cards in an SMP Redhat9 box, with 4 FXO ports each running thel latest Asterisk CVS. Users connect to this to share POTS lines using analog phones connected to Sipura SPA-2000 boxes. All works reasonably well, except every day (or more) a line will get locked up. No other way to describe it really -- the line just stops working, so if someone tries to dial out, and they happen to get assigned that port, all they hear is silence. The only way to unlock the port is to stop asterisk, and unload and reload the zaptel drivers (service zaptel restart), and then restart asterisk. I've seen other posts on the list referring to similar (same?) issues, but no resolutions. I have been staying current with the CVS version biweekly in hopes the problem will get resolved at some point. I have contacted Digium support (via email) about the issue. Has anyone made any headway with this problem? Thanks, Al -- Alok K. Dhir [EMAIL PROTECTED] Symplicity Corporation http://solutions.symplicity.com 703 351 6987 (w) | 703 351-6357 (f) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unreliable dtmf digit generation from tdm400p
I have the *exact* same problem. Please let me know if you have found any solution. Thanks! In my setup I have 2 of the TDM400P cards, with four FXO modules each. Al [EMAIL PROTECTED] wrote: I have a tdm400p 4 port fxo card which is not reliably creating the dtmf dialed digits when making a call. I have placed a linemans handset in monitor mode on the line and can hear that what the system reports it is dialing is not what the card is actually dialing. This happens about 25-50% of the time. The remaining time the digits dialed are correct and the call goes through properly. For example, I dial 5551212 == Spawn extension (default, 95551212, 1) exited non-zero on 'SIP/102-8da7' -- Executing Dial(SIP/102-07cb, Zap/2/5551212) in new stack -- Called 2/5551212 -- Zap/2-1 answered SIP/102-07cb The system logs that it's dialing 5551212 to channel zap/2.. great. Now when I actually listen to what the card is dialing, it doesn't dial 5551212 but something like 555212. I don't know what exactly it's dialing since I can't decode dtmf in my head, but it's clearly missing a digit or two. As a result, the telco comes back with a your call can't be completed because the full phone number wasn't dialed. I have a X100P which is also in the system which works just fine.. it never has this problem. This is a brand new card, and I only have this one, so I can't test with any others. Maybe it's defective?I've spent all day trying to troubleshoot this - I've tried different phone lines, even put the card into another box I built to try and troubleshoot. Always get the same intermittant problem. Also I've noticed in this testing that there is a slight pause before the last digit is dialed. This always occurs and I'm curious why it does this. Thanks for anyones help! Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alok K. Dhir [EMAIL PROTECTED] Symplicity Corporation http://solutions.symplicity.com 703 351 6987 (w) | 703 351-6357 (f) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
