Re: [asterisk-users] Asterisk Integration with Android device
Try media5 fone. I couldn't get 3cx to work on my iphone and tried about 7 different softfones. Media5 is the best by a long shot. Android is still in better and haven't tried it but if its anything like their iphone app it will be worth a look. There is a signup for the better at the website. let us know how you go. James - Original Message - From: bakko To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, August 24, 2011 11:48 PM Subject: Re: [asterisk-users] Asterisk Integration with Android device I think don't work with 2G network. Regards - Original Message - From: Gopal krishnan To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, August 24, 2011 4:01 PM Subject: [asterisk-users] Asterisk Integration with Android device Hi, I created a extension in Asterisk, the extension has been configured in Android softphone 3cx. When I tried to call from Andorid phone to some other IP extension which is registered in Asterisk, I am not able to hear the voice, when I check the asterisk log or wireshark there is only one way RTP traffic, from Android I am connecting to Asterisk via 2G GSM network. Any idea would be appreciated. Regards, Gopal -- _ -- 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
[asterisk-users] snom and srtp
Hi, I am running asterisk 1.8.5.0 and have compiled in the srtp module All but Snom phones are working. I have set the srtp tag on the snoms to 80 and RTP/SAVP to mandatory and they worked for a few hours. This morning all snoms are reporting this when trying to make a call (this is snom calling snom). -snip-- == Using SIP RTP CoS mark 5 -- Executing [1@default-outbound08:1] Dial(SIP/10002-0012, SIP/1,30) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/1 -- SIP/1-0013 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [1@default-outbound08:2] VoiceMail(SIP/10002-0012, 1,uj) in new stack [Aug 3 11:58:29] WARNING[9543]: res_srtp.c:384 ast_srtp_unprotect: SRTP unprotect: authentication failure [Aug 3 11:58:29] WARNING[9543]: res_srtp.c:384 ast_srtp_unprotect: SRTP unprotect: authentication failure -- SIP/10002-0012 Playing 'vm-theperson.g729' (language 'en') -- SIP/10002-0012 Playing 'digits/1.g729' (language 'en') -- SIP/10002-0012 Playing 'digits/0.g729' (language 'en') -- SIP/10002-0012 Playing 'digits/0.g729' (language 'en') -- SIP/10002-0012 Playing 'digits/0.g729' (language 'en') -- SIP/10002-0012 Playing 'digits/0.g729' (language 'en') sage*CLI Disconnected from Asterisk server [root@sage asterisk]# ---snip--- The interesting thing here is the call fails at this point and for some reason the cli disconnects when the call fails. Here is a call to a mobile which connects but the call dies in about 4 seconds --snip == Using SIP RTP CoS mark 5 -- Executing [0429835743@default-outbound08:1] Dial(SIP/10002-, SIP/private-sip/0429835743) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/private-sip/0429835743 -- SIP/private-sip-0001 is ringing -- SIP/private-sip-0001 answered SIP/10002- [Aug 3 12:06:05] WARNING[10146]: res_srtp.c:384 ast_srtp_unprotect: SRTP unprotect: authentication failure [Aug 3 12:06:05] WARNING[10146]: res_srtp.c:384 ast_srtp_unprotect: SRTP unprotect: authentication failure sage*CLI Disconnected from Asterisk server --snip I have done heaps of reading on SRTP unprotect error but cant really work it out from that. Q. should I try the patch mentioned below and forget about snoms doing 80 bit incription or should I persevere with making this work? thanks James ---snip--- Patch SRTP for 32bit SRTP have a cryptographic hash to check the integrity of the encrypted packets. It support two hash size: ● 32bit ● 80bit In order to properly fine tune SRTP for mobile networks and to have compatibility with PrivateGSM Enterprise we must use SRTP with hash at 32bit (HMAC_SHA1_32). Asterisk 1.8 by default does not announce in SDP both 32bit and 80bit, but only the 80bit version even if both are supported. This very small 1 line patch make Asterisk by default work with SRTP hash at 32bit . Download the patch for HMAC_SHA1_32 RTP crypto offer 48. wget http://sourceforge.net/projects/Asterisk-amr/files/1.8.0-rc2_crypto_offer.diff/download Apply the patch 49. cd Asterisk-1.8.0/ patch -p2 ../1.8.0-rc2_crypto_offer.diff Go to Asterisk-1.8.0/ folder50. cd .. Recompile Asterisk , 51. make ; make instal snip-- -- _ -- 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
