[asterisk-users] Dahdi QSIG with Tadiran Coral - not working
Hello All, we have Asterisk 1.8 connected to Tadiran Coral via RedFone Fonebridge. Our side is pri-net and Tadiran is pri-CPE, both sides have QSIG enabled. We are able to call internal user extensions in Tadiram and we are able to do outbound calls using Tadiran routes. We are even able to call outbound and perform DahdiSendCallReroutingFacility to internal user extension in Tadiram, thus joining both calls and freeing the channel in our Asterisk (this is a QSIG facility), as seen below. where 0019 is my mobile fone number. 1) Call my fone number 2) send it to extension 2@ura 3) extension 2@ura plays a message and calls DAHDISendCallreroutingFacility sending it to user extension 5501. 4) Fonebridge/Asterisk channel is released, pri show channels shows us that the call is not in our realm anymore. 5) Two connected legs can normally talk. [Apr 9 12:51:47] DEBUG[18196] sig_pri.c: sig_pri_request 1 [Apr 9 12:51:47] DEBUG[18196] sig_pri.c: CALLER NAME: NUM: [Apr 9 12:51:47] VERBOSE[18196] sig_pri.c: -- Requested transfer capability: 0x00 - SPEECH [Apr 9 12:51:54] VERBOSE[18199] pbx.c: -- Executing [2@ura:1] Playback(DAHDI/i1/0019-2, custom/ura/seguranca) in new stack [Apr 9 12:51:54] VERBOSE[18199] file.c: -- DAHDI/i1/0019-2 Playing 'custom/ura/seguranca.slin' (language 'en') [Apr 9 12:51:59] VERBOSE[18199] pbx.c: -- Executing [2@ura:2] DAHDISendCallreroutingFacility(DAHDI/i1/0019xxx-2, 5501) in new stack [Apr 9 12:51:59] WARNING[18199] chan_dahdi.c: Callrerouting Facility without original called number argument [Apr 9 12:51:59] NOTICE[18199] chan_dahdi.c: Callrerouting Facility without diversion reason argument, defaulting to unknown [Apr 9 12:51:59] VERBOSE[18199] pbx.c: == Spawn extension (ura, 2, 2) exited non-zero on 'DAHDI/i1/0019-2' [Apr 9 12:51:59] VERBOSE[18199] chan_dahdi.c: -- Hungup 'DAHDI/i1/0019-2' But, inbound to our system, Tadiran can send us absolutely nothing. I have set pri intense debug span 1 and cannot see anything happening coming from Tadiram. No log at all, only exchange of regular protocol messages. My conclusion is that Tadiram tech guys are doing something wrong when trying to send us a call. Is it possible that, even having no log messages, no error messages, and after beeing able to do the mentioned above, that the problem is in our side? Thanks, Eduardo -- -- _ -- 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] Dahdi QSIG with Tadiran Coral - not working
Hi Eduardo, we use redfone in most of our projects. If you have enabled intense debug on that particular span, you should see some logs coming in from your Tadiran. If you don't see any, the problem is from Tadiran side, I highly suspect. On Sat, Apr 14, 2012 at 11:19 PM, Eduardo Pimenta e...@alertbrasil.com.brwrote: Hello All, we have Asterisk 1.8 connected to Tadiran Coral via RedFone Fonebridge. Our side is pri-net and Tadiran is pri-CPE, both sides have QSIG enabled. We are able to call internal user extensions in Tadiram and we are able to do outbound calls using Tadiran routes. We are even able to call outbound and perform DahdiSendCallReroutingFacility to internal user extension in Tadiram, thus joining both calls and freeing the channel in our Asterisk (this is a QSIG facility), as seen below. where 0019 is my mobile fone number. 1) Call my fone number 2) send it to extension 2@ura 3) extension 2@ura plays a message and calls DAHDISendCallreroutingFacility sending it to user extension 5501. 4) Fonebridge/Asterisk channel is released, pri show channels shows us that the call is not in our realm anymore. 5) Two connected legs can normally talk. [Apr 9 12:51:47] DEBUG[18196] sig_pri.c: sig_pri_request 1 [Apr 9 12:51:47] DEBUG[18196] sig_pri.c: CALLER NAME: NUM: [Apr 9 12:51:47] VERBOSE[18196] sig_pri.c: -- Requested transfer capability: 0x00 - SPEECH [Apr 9 12:51:54] VERBOSE[18199] pbx.c: -- Executing [2@ura:1] Playback(DAHDI/i1/0019-2, custom/ura/seguranca) in new stack [Apr 9 12:51:54] VERBOSE[18199] file.c: -- DAHDI/i1/0019-2 Playing 'custom/ura/seguranca.slin' (language 'en') [Apr 9 12:51:59] VERBOSE[18199] pbx.c: -- Executing [2@ura:2] DAHDISendCallreroutingFacility(DAHDI/i1/0019xxx-2, 5501) in new stack [Apr 9 12:51:59] WARNING[18199] chan_dahdi.c: Callrerouting Facility without original called number argument [Apr 9 12:51:59] NOTICE[18199] chan_dahdi.c: Callrerouting Facility without diversion reason argument, defaulting to unknown [Apr 9 12:51:59] VERBOSE[18199] pbx.c: == Spawn extension (ura, 2, 2) exited non-zero on 'DAHDI/i1/0019-2' [Apr 9 12:51:59] VERBOSE[18199] chan_dahdi.c: -- Hungup 'DAHDI/i1/0019-2' But, inbound to our system, Tadiran can send us absolutely nothing. I have set pri intense debug span 1 and cannot see anything happening coming from Tadiram. No log at all, only exchange of regular protocol messages. My conclusion is that Tadiram tech guys are doing something wrong when trying to send us a call. Is it possible that, even having no log messages, no error messages, and after beeing able to do the mentioned above, that the problem is in our side? Thanks, Eduardo -- -- _ -- 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 -- Regards, Arstan Jusupov -- _ -- 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] dahdi versions before 2.5 compilation error and ubuntu
Hi All; It look like DAHDI versions that before 2.5 have a problem to be compiled on ubuntu, can someone check below and advise me how to fix this? The output of the uname -a is: Linux House 3.0.0-17-server #30-Ubuntu SMP Thu Mar 8 22:15:30 UTC 2012 x86_64 x86_64 x86_64 GNU/Linux I am trying to install DAHDI on ubuntu and I faced a problem if I am compiling versions before 2.5, but if I tried 2.5 and newer then it is working fine. The error is: make[2]: Entering directory `/usr/src/linux-headers-3.0.0-17-server' CC [M] /usr/src/dahdi-linux-complete-2.4.0+2.4.0/linux/drivers/dahdi/dahdi-base.o /usr/src/dahdi-linux-complete-2.4.0+2.4.0/linux/drivers/dahdi/dahdi-base.c:52:28: fatal error: linux/smp_lock.h: No such file or directory compilation terminated. make[3]: *** [/usr/src/dahdi-linux-complete-2.4.0+2.4.0/linux/drivers/dahdi/dahdi-base.o] Error 1 make[2]: *** [_module_/usr/src/dahdi-linux-complete-2.4.0+2.4.0/linux/drivers/dahdi] Error 2 make[2]: Leaving directory `/usr/src/linux-headers-3.0.0-17-server' make[1]: *** [modules] Error 2 make[1]: Leaving directory `/usr/src/dahdi-linux-complete-2.4.0+2.4.0/linux' make: *** [all] Error 2 Regards Bilal -- _ -- 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] dahdi versions before 2.5 compilation error and ubuntu
On Sat, Apr 14, 2012 at 01:36:02PM -0700, bilal ghayyad wrote: Hi All; It look like DAHDI versions that before 2.5 have a problem to be compiled on ubuntu, can someone check below and advise me how to fix this? The output of the uname -a is: Linux House 3.0.0-17-server #30-Ubuntu SMP Thu Mar 8 22:15:30 UTC 2012 x86_64 x86_64 x86_64 GNU/Linux I am trying to install DAHDI on ubuntu and I faced a problem if I am compiling versions before 2.5, but if I tried 2.5 and newer then it is working fine. Why not use DAHDI 2.5.0.2 then? What reason do you have for staying on 2.4? The error is: make[2]: Entering directory `/usr/src/linux-headers-3.0.0-17-server' CC [M] /usr/src/dahdi-linux-complete-2.4.0+2.4.0/linux/drivers/dahdi/dahdi-base.o /usr/src/dahdi-linux-complete-2.4.0+2.4.0/linux/drivers/dahdi/dahdi-base.c:52:28: fatal error: linux/smp_lock.h: No such file or directory compilation terminated. make[3]: *** [/usr/src/dahdi-linux-complete-2.4.0+2.4.0/linux/drivers/dahdi/dahdi-base.o] Error 1 make[2]: *** [_module_/usr/src/dahdi-linux-complete-2.4.0+2.4.0/linux/drivers/dahdi] Error 2 make[2]: Leaving directory `/usr/src/linux-headers-3.0.0-17-server' make[1]: *** [modules] Error 2 make[1]: Leaving directory `/usr/src/dahdi-linux-complete-2.4.0+2.4.0/linux' make: *** [all] Error 2 You are attempting to build dahdi-linux 2.4.0. The problem that you are describing was addressed in commit r9721 dahdi: Experimentally remove dependency on the Big Kernel Lock. [1] and is in 2.4.1. So I believe you should be good to go with dahdi-linux 2.4.1.2. [1] http://svnview.digium.com/svn/dahdi?view=revisionrevision=9721 Keep in mind that the 2.4 branch isn't actively maintained so whether or not it continues to get back ported patches as the linux kernel interfaces change is going to be ad-hoc at best. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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] Unable to create channel of type 'IAX2' (cause 20 - Unknown)
I have two asterisk servers connected via iax.
home_server = IAX2 = clinic_server
I'm just testing, calling from home_server via clinic_server but I'm
getting an error message:
Call gets through to clinic_server but will not call back.
Dial(IAX2/home_server-957, IAX2/home_server/218,30,rw) in new stack
[Apr 14 16:59:45] WARNING[27870]: app_dial.c:2218 dial_exec_full: Unable to
create channel of type 'IAX2' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [219@extensions:2] Hangup(IAX2/home_server-957, ) in new stack
This is on clinic_server:
exten = 218,1,Dial(IAX2/home_server/${EXTEN},30,rw)
exten = 218,n,Hangup
exten = 219,1,Dial(IAX2/home_server/218,30,rw)
exten = 219,n,Hangup
When somebody in a clinic server dials: 218 my phone in home_server is ringing
OK
But when I call extension 219 from home server it should dial my 'home_server'
218 and it does but it gives me an error message:
app_dial.c:2218 dial_exec_full: Unable to create channel of type 'IAX2' (cause
20 - Unknown)
--
Joseph
--
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Re: [asterisk-users] Unable to create channel of type 'IAX2' (cause 20 - Unknown)
I forgot to add:
clinic-amd*CLI iax2 show peers
Name/UsernameHost Mask Port Status
home_server (null) (D) 255.255.255.255 0 Unmonitored
iaxy-322/iaxy-3 (null) (D) 255.255.255.255 0 Unmonitored
--
Joseph
On 04/14/12 17:09, Joseph wrote:
I have two asterisk servers connected via iax.
home_server = IAX2 = clinic_server
I'm just testing, calling from home_server via clinic_server but I'm
getting an error message:
Call gets through to clinic_server but will not call back.
Dial(IAX2/home_server-957, IAX2/home_server/218,30,rw) in new stack
[Apr 14 16:59:45] WARNING[27870]: app_dial.c:2218 dial_exec_full: Unable to
create channel of type 'IAX2' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [219@extensions:2] Hangup(IAX2/home_server-957, ) in new stack
This is on clinic_server:
exten = 218,1,Dial(IAX2/home_server/${EXTEN},30,rw)
exten = 218,n,Hangup
exten = 219,1,Dial(IAX2/home_server/218,30,rw)
exten = 219,n,Hangup
When somebody in a clinic server dials: 218 my phone in home_server is ringing
OK
But when I call extension 219 from home server it should dial my 'home_server'
218 and it does but it gives me an error message:
app_dial.c:2218 dial_exec_full: Unable to create channel of type 'IAX2' (cause
20 - Unknown)
--
Joseph
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Can't make Asterisk send authentication to remote peer on INVITE
172.16.0.1 is not sending the authentication details to 172.16.0.2 when 172.16.0.2 responds with 401. On Sat, Apr 14, 2012 at 9:30 PM, Ben WIlliams bwilliams+aster...@jadeworld.com wrote: This is a really simple problem that I just can't get to work. There are two Asterisk servers with the following sip user and peer. When a call is attempted, Asterisk is not sending authentication details in response to the 401. Note, if the secret is blank on 172.16.0.2 test, the INVITE succeeds. on 172.16.0.2: [test] type=friend secret=abcde host=dynamic context=demo on 172.16.0.1 : [natty] type=peer host=172.16.0.2 fromuser=test secret=abcde originate SIP/natty/1234568 extension 200 == Using SIP RTP CoS mark 5 Audio is at 172.16.0.1 port 19486 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 172.16.0.2:5060: INVITE sip:1234568@172.16.0.2 SIP/2.0 Via: SIP/2.0/UDP 172.16.0.1:5066;branch=z9hG4bK59f50e30;rport Max-Forwards: 70 From: asterisk sip:test@172.16.0.1:5066;tag=as1689b2b6 To: sip:1234568@172.16.0.2 Contact: sip:test@172.16.0.1:5066 Call-ID: 2353cf0e59596e285c684b44220f8915@172.16.0.1 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2.1 Date: Sat, 14 Apr 2012 09:10:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 290 v=0 o=root 1594270426 1594270426 IN IP4 172.16.0.1 s=Asterisk PBX 1.6.2.9-2ubuntu2.1 c=IN IP4 172.16.0.1 t=0 0 m=audio 19486 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- --- SIP read from UDP:172.16.0.2:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.16.0.1:5066;branch=z9hG4bK59f50e30;received=172.16.0.1;rport=5066 From: asterisk sip:test@172.16.0.1:5066;tag=as1689b2b6 To: sip:1234568@172.16.0.2;tag=as1a6c2364 Call-ID: 2353cf0e59596e285c684b44220f8915@172.16.0.1 CSeq: 102 INVITE Server: Asterisk PBX 1.6.2.9-2ubuntu2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=7a03a1d3 Content-Length: 0 - --- (11 headers 0 lines) --- Transmitting (no NAT) to 172.16.0.2:5060: ACK sip:1234568@172.16.0.2 SIP/2.0 Via: SIP/2.0/UDP 172.16.0.1:5066;branch=z9hG4bK59f50e30;rport Max-Forwards: 70 From: asterisk sip:test@172.16.0.1:5066;tag=as1689b2b6 To: sip:1234568@172.16.0.2;tag=as1a6c2364 Contact: sip:test@172.16.0.1:5066 Call-ID: 2353cf0e59596e285c684b44220f8915@172.16.0.1 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2.1 Content-Length: 0 --- [Apr 14 21:10:38] NOTICE[31158]: chan_sip.c:17975 handle_response_invite: Failed to authenticate on INVITE to 'asterisk sip:test@172.16.0.1:5066;tag=as1689b2b6' -- _ -- 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
