[asterisk-users] WARNING[17634]: chan_sip.c:3906 __sip_xmit: sip_xmit of 0x7fe394006590 (len 609) to 83.78.150.198:60709 returned -2: Success

i'm using asterisk with tls but always get WARNING[17634]: chan_sip.c:3906 __sip_xmit: sip_xmit of 0x7fe394006590 (len 609) to 83.78.150.198:60709 returned -2: Success whats wrong there? Best Regards Jakob signature.asc Description: OpenPGP digital signature --

[asterisk-users] SRTP only from asterisk to extention possible

Hello, trying to implement srtp with already working tls i somehow stuck with srtp. If the extension has successfully registered a call from asterisk to that extension works fine. But the other way round nothing happens. [Aug 13 14:54:16] WARNING[31053]: chan_sip.c:3906 __sip_xmit: sip_xmit of

[asterisk-users] Asterisk 11.11 with TCP/TLS SRTP and Grandstream gxp1450 not working

Hey there i'm trying to get an Asterisk 11.11 with encryption working with my Grandstream phones. But i stuck. To avoid NAT problems i'm using IPv6 Just with TCP/TLS it's working fine. Only the SRTP funktion is not working. Asterisk tells me WARNING[6938]: chan_sip.c:3906 __sip_xmit: sip_xmit

Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

Hi, changing faxdetect=cng and t38pt_udptl=no helped making it work. Thanks Am 03.02.2014 11:57, schrieb Larry Moore: On 3/02/2014 3:34 PM, Jakob-Matthias Böttger wrote: . . . [sipcall.ch] type=peer insecure=invite defaultuser=123456789 fromuser=123456789 fromdomain=voipdomain.com secret

Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

]: chan_sip.c:10497 process_sdp: Insufficient information in SDP (c=)... and then the fax session starts recording data Am 03.02.2014 12:34, schrieb Larry Moore: On 3/02/2014 7:15 PM, Jakob-Matthias Böttger wrote: Hi, changing faxdetect=cng and t38pt_udptl=no helped making it work. Hmm, the fax

Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

Am 03.02.2014 12:56, schrieb Larry Moore: On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote: as He is describing it he should actually provide t.38. but i don't know why it is not working thus im now getting Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10353 process_sdp: Failed

Re: [asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

Am 03.02.2014 13:20, schrieb Jakob-Matthias Böttger: Am 03.02.2014 12:56, schrieb Larry Moore: On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote: as He is describing it he should actually provide t.38. but i don't know why it is not working thus im now getting Feb 3 12:32:55] WARNING[9942

[asterisk-users] Incoming Fax Issue with Asterisk 11.7 and Digium Fax

Hi, im using a Asterisk Server which is not behind NAT. First i had problems with the fax detection. But this is now solved after adding a wait(2) at the correct place. But i'm still unable to receive a fax due to res_rtp_asterisk.c:3548 ast_rtp_read: RTP Read too short after the Fax session

Re: [asterisk-users] Asterisk Fax detection *11.7

Hello, now i added directmedia=no disallow=all allow=ulaw allow=alaw and i changed the caninvite part to canreinvite. Now the faxdetection is working well. But now, after the faxsession has started, i'm getting res_rtp_asterisk.c:3548 ast_rtp_read: RTP Read too short as error. Regards

[asterisk-users] Asterisk Fax detection *11.7

Hello everybody I'm trying to enable the Digium res_fax app at my *11.7 Server. a fax show stats comes up with FAX Statistics: --- Current Sessions : 0 Reserved Sessions: 0 Transmit Attempts: 0 Receive Attempts : 1 Completed FAXes : 1 Failed FAXes : 1

Re: [asterisk-users] Asterisk Fax detection *11.7

Hi The log i've posted == Using SIP VIDEO CoS mark 6 == Using SIP RTP CoS mark 5 -- Executing [12345678912 tel:%5B12345678912@from-sip:1] Answer(SIP/abcde-0016, ) in new stack 0x7fd11404cd00 -- Probation passed - setting RTP source address to 123.456.789.123:17108 --

Re: [asterisk-users] Asterisk Fax detection *11.7

it. Leandro 2014/1/21 Jakob-Matthias Böttger ja...@j-mb.de mailto:ja...@j-mb.de Hi The log i've posted == Using SIP VIDEO CoS mark 6 == Using SIP RTP CoS mark 5 -- Executing [12345678912 tel:%5B12345678912@from-sip:1] Answer(SIP/abcde-0016, ) in new stack

[asterisk-users] Queue Member login from IAX trunk

Hello i got two Asterisk servers connected with IAX2. At server 1 i hosted a queue named support. Now i want the Agent to login from SIP Phones Connected to server B. Therefore i wrote a extension like exten = 105,1,Authenticate(1234) exten = 105,n,AddQueueMember(support,,) exten =

Re: [asterisk-users] Queue Member login from IAX trunk

line doing in SET() !!? Please explain that. Regards, Sammy On Wed, Jul 4, 2012 at 2:32 PM, Jakob-Matthias Böttger ja...@j-mb.de mailto:ja...@j-mb.de wrote: Hello i got two Asterisk servers connected with IAX2. At server 1 i hosted a queue named support. Now i want the Agent

[asterisk-users] app_swift 3 and asterisk 1.8.13.0 fails with undefined symbol: swift_port_close

Hi, i am trying to install the just from git cloned app_swift version. Compiling works fine. Install as well. But if i try to load the module at Asterisk it fails with. Command 'module load app_swift.so ' failed. [Jun 20 11:29:51] WARNING[24217]: loader.c:460 load_dynamic_module: Error

Re: [asterisk-users] app_swift 3 and asterisk 1.8.13.0 fails with undefined symbol: swift_port_close

20, 2012, at 4:41 AM, Jakob-Matthias Böttger wrote: Hi, i am trying to install the just from git cloned app_swift version. Compiling works fine. Install as well. But if i try to load the module at Asterisk it fails with. Command 'module load app_swift.so ' failed. [Jun 20 11:29:51] WARNING