[Asterisk-Users] g729 recording on asterisk using g729 enabled phone
hi, i have installed asterisk on my system and using only g729 enabled phones. from what i understand, we would not be needing any g729 licenses as all my voicemail prompts are also in g729 and asterisk is not doing any transcoding. when i use the voicemail function to record, the message is not recorded (0 byte file is created) and it gives the following errors - unable to convert from g729 to slin in my voicemail.conf, the format is set as format=g729 and in sip.conf, disallow=all allow=g729 allow=ulaw allow=alaw the asterisk version is 1.0.5 what might be the reason ? regards, tulika _ _ Formula One fan? http://server1.msn.co.in/sp05/tataracing/ Get news, wallpapers and photos of Narain Karthikeyan. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk crashes
hi !
i have the following in my extensions.conf
exten = 2000,1,Wait(60)
exten = 2000,2,Hangup
When i dial '2000' from my phone, I see 'Wait' being called. After 60 secs,
I also se 'Hangup' being called. If I hangup the phone line before 60 secs
are over ('Wait' command is probably interrupted in this case), asterisk
crashes with segmentation fault.
Due to this problem, my 'campon' feature causes asterisk to crash often.
does anyone have an idea as to what this problem might be ?
tulika
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RE: [Asterisk-Users] Call Transfer using SIP clients
call transfer works for me fine without any additions in features.conf
by simply using Dial(SIP/${EXTEN},20,tT)
and pressing #number to be transfered to
this works both from caller as well as callee.
tulika
From: Frank Schoep [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Call Transfer using SIP clients
Date: Mon, 4 Jul 2005 16:11:13 +0200
Hello all,
First of all, let me apologize about the length of this message, but I
suppose
it was necessary to include the details.
I've spent quite some time already trying to get the call transfer function
to
work on my Asterisk installation. Let me first describe the general
situation
of the setup I am using, so you might be able to pinpoint the cause of the
problem.
I'm currently using Asterisk CVS as of July 4th 2005. The only means of
communication at the moment is the XTen X-Lite SIP Client, I already added
the following entries to my sip.conf configuration file:
[frank]
canreinvite=no
type=friend
secret=frank
username=frank
nat=yes
host=dynamic
[test]
canreinvite=no
type=friend
secret=test
username=test
nat=yes
host=dynamic
The SIP setup is working without a problem, the X-Lite application
correctly
registers the users and I can set up calls between them. I've also tested
queues and they work without a problem, too. Next up is my extensions
configuration, of which the interesting section now looks like this:
[default]
include = general ; longshot, added out of desparation
include = parkedcalls ; longshot, added out of desparation
include = featuremap ; longshot, added out of desparation
exten = 800,1,Answer
exten = 800,2,Dial(SIP/frank,20,tT)
exten = 800,3 Hangup
exten = 802,1,Answer
exten = 802,2,Dial(SIP/test,20,tT)
exten = 802,3 Hangup
Notice the inclusion of several contexts that should or would have to be
defined in the features configuration. My features.conf looks something
like
this, I trimmed the 'general' section for brevity:
[general]
; (trimmed) default options
[featuremap]
blindxfer = #1 ; Blind transfer
disconnect = *0 ; Disconnect
automon = *1 ; One Touch Record
atxfer = *2 ; Attended transfer
My testing scenario starts as follows:
- log in both X-Lite SIP clients
- from the 'test' phone, call extension 800
- on X-Lite client 'frank' accept the call
- talk to eachother
At this point I want to transfer to call to another extension, also defined
in
sip.conf but unlisted here. The problem is that nothing happens when I
press the #1 or *2 keys in the 'frank' X-Lite client. I also tested
these
key combinations on the 'test' X-Lite client during the call, but that also
had not effect.
I searched the web and the mailing list archive for a solution, and if I
recall correctly, someone stated that call transfer is only available for
calls originating from the PSTN. Is this correct, also in regard of the
current version of Asterisk? Has anyone got an idea how to get call
transfer
to work?
One thing I tried was to change the DTMF settings in the clients, so they
are
sent in-band, but this also didn't help. Should I revert this option?
Thanks in advance for your time and patience.
Sincerely,
Frank Schoep
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RE: [Asterisk-Users] Problem with asterisk+gnugk
hi ! i have used pre-compiled gnugk version 2.2.1-2 with the mentioned versions of pwlib and openh323 (1.5.2 and 1.12.2) respectively and it works fine. tulika From: [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Problem with asterisk+gnugk Date: Tue, 31 May 2005 10:14:25 +0300 Hi! I'm trying to build gnugk with asterisk. Asterisk is working well with chan_h323 built with needed PWlib v.1.5.2 and open H.323 v.1.12.2. But gnugk' s installing instructions says that I need latest PWlib(1.17.1) and openh323 to get gnugk work. Now, with installed pwlib and openh323 gnugk's compiling fails and I get error 1. Do you have any working solutions with asterisk and gnugk and what are needed version numbers which you use to get then work together? Thanks in advance! This mail sent through L-secure: http://www.l-secure.net/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Get yourself a brand new Mobile. http://adfarm.mediaplex.com/ad/ck/4686-26272-10936-378?ck=BuyNewMobile Find,Compare BUY IT NOW on eBay.in! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sipsak with asterisk
i am attaching the trace of the sipsak error when run with the command below. # sipsak -UI -a password -s sip:[EMAIL PROTECTED]:5060 - warning: ignoring -i option when in usrloc mode fqdnhostname: 127.0.0.1 username: 985389744 domainname: 203.197.212.211:5060 request: REGISTER sip:203.197.212.211:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:1026;rport From: sip:[EMAIL PROTECTED]:5060;tag=3b60505e To: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER Contact: sip:[EMAIL PROTECTED]:1026 Expires: 15 Content-Length: 0 Max-Forwards: 70 User-Agent: sipsak 0.8.12 registering user 985389744... ignoring provisinal response authorizing received: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 127.0.0.1:1026;received=203.197.212.211;rport=1026 From: sip:[EMAIL PROTECTED]:5060;tag=3b60505e To: sip:[EMAIL PROTECTED]:5060;tag=as72f0756d Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=asterisk, nonce=5556cf14 Content-Length: 0 registering user 985389744... ignoring provisinal response OK SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:1026;received=203.197.212.211;rport=1026 From: sip:[EMAIL PROTECTED]:5060;tag=3b60505e To: sip:[EMAIL PROTECTED]:5060;tag=as72f0756d Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 15 Contact: sip:[EMAIL PROTECTED]:1026;expires=15 Date: Thu, 26 May 2005 04:03:58 GMT Content-Length: 0 username: 985389744 domainname: 203.197.212.211:5060 ack: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:1026;rport From: sip:[EMAIL PROTECTED]:1026;tag=4c37a25f To: sip:[EMAIL PROTECTED]:5060;tag=11415721137 Call-ID: [EMAIL PROTECTED] CSeq: 2 ACK Content-Length: 0 Max-Forwards: 70 User-Agent: sipsak 0.8.12 reply: SIP/2.0 200 OK From: sip:[EMAIL PROTECTED]:1026;tag=4c37a25f To: sip:[EMAIL PROTECTED]:5060;tag=11415721137 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE Content-Length: 0 User-Agent: sipsak 0.8.12 request: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:1026;rport From: sip:[EMAIL PROTECTED]:1026;tag=4c37a25f To: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE Content-Length: 0 Contact: sip:[EMAIL PROTECTED]:1026 Subject: DONT ANSWER this test call! Max-Forwards: 70 User-Agent: sipsak 0.8.12 inviting user 985389744... authorizing received: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 127.0.0.1:1026;received=203.197.212.211;rport=1026 From: sip:[EMAIL PROTECTED]:1026;tag=4c37a25f To: sip:[EMAIL PROTECTED]:5060;tag=as680d9091 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=0469ee64 Content-Length: 0 error: could not find To in the reply -- tulika From: Tulika Pradhan [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] sipsak with asterisk Date: Fri, 13 May 2005 05:55:01 + i am using sipsak to test asterisk. i use the command $ sipsak -U -I -s sip:[EMAIL PROTECTED] -vv -x 1800 -e3 and i get the message SIP/2.0 407 Proxy Authentication Required as a response to INVITE message (REGISTER was successful) and error: could not find To in the reply does anyone have some idea as to what is missing ? tulika _ Kareena or Rani? Saif or SRK? http://server1.msn.co.in/sp05/iifa/ Rock your vote now at IIFA. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ The MSN Gamezone! http://www.msn.co.in/gamezone Ready for the challenge? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk crashes with sipp
with asterisk running, if i call sipp ip address -s 9111 -d 6 -r 20 -t un -sn uac -m 60 all the calls get set up, and after a minute when asterisk receives the 1st BYE from uac, it responds with 200 OK and then crashes. If i restart asterisk, all the calls get terminated properly. in the extensions, i have [default] exten = 9111222,1,Answer exten = 9111222,3,Wait(600) exten = 9111222,4,Hangup please help as i am unable to continue with any load tests ! tulika _ Kareena or Rani? Saif or SRK? http://server1.msn.co.in/sp05/iifa/ Rock your vote now at IIFA. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sipsak with asterisk
i am using sipsak to test asterisk. i use the command $ sipsak -U -I -s sip:[EMAIL PROTECTED] -vv -x 1800 -e3 and i get the message SIP/2.0 407 Proxy Authentication Required as a response to INVITE message (REGISTER was successful) and error: could not find To in the reply does anyone have some idea as to what is missing ? tulika _ Kareena or Rani? Saif or SRK? http://server1.msn.co.in/sp05/iifa/ Rock your vote now at IIFA. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIPP with asterisk
calls get setup with the config below but in the stats all calls are failed. BYE are sent from uac to asterisk but after no more than 1 response is receieved. All calls show to be failed in stats printed by sipp (uac). isn't the uas part of sipp supposed to received 200 messages after BYE has been sent. That is not coming into picture. From: Tim Connolly [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] SIPP with asterisk Date: Thu, 5 May 2005 00:41:25 -0500 How about: exten = 9111222,1,answer exten = 9111222,2,wait(10) exten = 9111222,3,Hangup -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tulika Pradhan Sent: Wednesday, May 04, 2005 11:29 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIPP with asterisk i am trying to do load testing on asterisk using sipp testing tool. i am able to send invite requests to asterisk by using sipp -sn uac ip address -s 9111222 -d 1 -r 10 i am also running sipp -sn uas on the same box but no message arrives on uas part. and asterisk returns error while dialing. Unable to create channel of type SIP the extensions.conf has exten = 9111222,1,Dial(SIP/9111222) exten = 9111222,2,Hangup what config changes should i make to answer the calls landing on asterisk ? tulika _ Insta predictions! http://www.astroyogi.com/newmsn/astroshopping/astrologerservices/express.asp Get your answers in 48 hours! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Bought a New Cellphone? http://adfarm.mediaplex.com/ad/ck/4686-26272-10936-265?ck=Register Sell your old one for a Great Price in eBay! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIPP with asterisk
i am trying to do load testing on asterisk using sipp testing tool. i am able to send invite requests to asterisk by using sipp -sn uac ip address -s 9111222 -d 1 -r 10 i am also running sipp -sn uas on the same box but no message arrives on uas part. and asterisk returns error while dialing. Unable to create channel of type SIP the extensions.conf has exten = 9111222,1,Dial(SIP/9111222) exten = 9111222,2,Hangup what config changes should i make to answer the calls landing on asterisk ? tulika _ Insta predictions! http://www.astroyogi.com/newmsn/astroshopping/astrologerservices/express.asp Get your answers in 48 hours! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Taking asterisk out of the media path - SIP - howis it achieved
the following are necessary conditions for asterisk out of mediia path - - the codecs on both endpoints of the call should be the same. - the technology on both endpoints should be the same. - canreinvite=yes - no 't', 'T', L, conference, transfer flags in Dial command tulika From: David John Walsh [EMAIL PROTECTED] Reply-To: David John Walsh [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [Asterisk-Users] Taking asterisk out of the media path - SIP - howis it achieved Date: Mon, 2 May 2005 21:41:49 +0100 Hello How do you make asterisk stay out of the media stream? i.e once I set a call up between two parties, even if asterisk fell over the call would continue (in the same way a HLR on a mobile network works) I understand that many features will be lost if I do this, but all that I need seems to be supported by the end user hardware. incidentally I have tried canreinvite=yes, doesn't seem to work. I have also tried removing any flags in the dial() command Thank you for any information. David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Bought a New Cellphone? http://adfarm.mediaplex.com/ad/ck/4686-26272-10936-265?ck=Register Sell your old one for a Great Price in eBay! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] signaling during a call
I am using Asterisk with SIP phones. is it possible to press a key during a conversation and get asterisk to do something? Like the # key, but I would like asterisk to take other actions instead of transfering. tulika _ Your [EMAIL PROTECTED] Spaces! http://www.msn.co.in/spaces Blogs, albums, music lists. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
