[asterisk-users] Hangupcause on DAHDI 2.4.9-svn-r9328 channels - Asterisk 1.4.36
Hello, I face a problem on some dahdi incoming calls. Hardware is Xorcom with Elastix 1.6.2.27/Asterisk 1.4.36/DAHDI 2.4.9-svn-r9328 inside. Setup is 3 incoming BRI (euroisdn), ringing phones are 3xSNOM320, 4xSNOM300 and 4xFXS phones. On this calls, phones are ringing and when picked up, nobody on the other end. Other phones are still ringing and same behavior when trying to pickup. No need to say that *8 doesn't work better. I checked one of those call and found that the callee hanged up before someone picked up the call. What I have in logs (debug) logger.c: -- Channel 0/1, span 3 got hangup, cause 111 rtp.c: Channel 'DAHDI/7-1' has no RTP, not doing anything further app_dial.c: Exiting with DIALSTATUS=CANCEL further logger.c: Protocol Discriminator: Q.931 (8) len=8 logger.c: TEI=82 Call Ref: len= 1 (reference 1/0x1) (Sent to originator) logger.c: Message Type: RELEASE COMPLETE (90) logger.c: [08 02 81 d1] logger.c: Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) logger.c: Ext: 1 Cause: Invalid call reference value (81), class = Invalid message (e.g. parameter out of range) (5) ] Questions are: why are phones continuing to ring after the call had been canceled? What are hangup cause 111 and Invalid message/parameter out of range. Thanks for any hint -- Daniel -- _ -- 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] dialplan problem : not including context
Le 13/01/2012 14:32, Jonas Kellens a écrit :
On 01/13/2012 02:23 PM, Doug Lytle wrote:
Jonas Kellens wrote:
I have the following in dialplan :
[TrunkAccounts]
dialplan show TrunkAccounts
Make sure the sort order is what you're expecting.
Doug
Hello,
The order is correct for as far as I'm sure.
[TrunkAccounts]
exten = 32380837,1,GoTo(01,32380837,1)
exten = 32380838,1,GoTo(01,32380838,1)
exten = 32380839,1,GoTo(01,32380839,1)
[CheckOnNet]
include = TrunkAccounts
exten = _321[0-3],1,GoTo(context1,${EXTEN},1)
exten = 3214,1,GoTo(context2, ${EXTEN} ,1)
exten = _.,1,NoOp()
exten = _.,n,Return()
Are you sure about your _. exten? Typo in the mail? It means 9
and more digits but your extensions are 8 digits ...
Include are always treated *after* context command. If _. is
right, something is wrong with Asterisk as it should treat
TrunkAccounts. If _XXX. (8 digits or more) is what you have in
yourdialplan, than the behavior of Asterisk is OK
Try
[TrunkAccounts]
exten = 32380837,1,GoTo(01,32380837,1)
exten = 32380838,1,GoTo(01,32380838,1)
exten = 32380839,1,GoTo(01,32380839,1)
[TrunkNotTreated]
exten = _.,1,NoOp()
exten = _.,n,Return()
[CheckOnNet]
include = TrunkAccounts
include = TrunkNotTreated
exten = _321[0-3],1,GoTo(context1,${EXTEN},1)
exten = 3214,1,GoTo(context2, ${EXTEN} ,1)
[...]
--
Daniel
--
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Re: [asterisk-users] how to stop hacking of my server
Le 27/12/2011 16:04, Tim Nelson a écrit : - Original Message - On Mon, Dec 26, 2011 at 11:54 PM, virendra bhati virbh...@gmail.com wrote: Hi list someone is trying to hack my server . Is there any way by whcih I can stop hacking of my server except iptables ? [...] Odd nobody else mentioned it yet, so I'll do it... Check out fail2ban. [...] He said except iptables. fail2ban is iptables related ;-) -- Daniel -- _ -- 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] Populate CDR issues
Le 06/12/2011 10:16, Harel Cohen a écrit : Hello Everyone, Hi Harel I didn’t get a reply to my problem below so I’m posting again just in case someone who might be able to help missed my previous post. Thank You… Please take a look at issue ASTERISK-18875 https://issues.asterisk.org/jira/browse/ASTERISK-18875 [...] -- Daniel -- _ -- 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] Issue with Polycom SPIP650 and its sidecar [SOLVED]
Le 01/12/2011 13:44, Olivier a écrit : [...] I still can explain myself why a PoE switch (a Linksys SRW224P) would succeed or fail to deliver power to a plugged IP phone, given that only a couple of Polycom phones are using this switch a power source. I think your switch deliver a max value of power per port, the phone and side-car take are just on this limit. -- Daniel -- _ -- 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] 2 same sip extension number on 2 asterisk - call not passing on certain condition
Hi list, something crazy here. 2 asterisk on 2 different place (1.4 and 1.8) both having an extension [115], one as type peer (caller side 1.4) and one as friend (callee side 1.8). Phones from both location connect to Asterisk from LAN. Router are Linux boxes. Connection between the 2 sites is done like this: On the callee side [115] ;callee type=friend host=dynamic secret=otherSecret context=local nat=no canreinvite=no qualify=no dtmfmode=rfc2833 allow=all call-limit=1 busy-level=1 allow=all [Caller] type=peer host=voip1.domain.net deny=0.0.0.0/0.0.0.0 permit=xxx.xxx.xxx.xxx context=myAccess disallow=all allow=all nat=yes insecure=port,invite On the caller side [115] type=peer username=115 secret=blabla context=local host=dynamic nat=yes canreinvite=no dtmfmode=auto disallow=all allow=jpeg,png,h263,h263p,h264,alaw,ulaw callgroup=1 pickupgroup=1 insecure=invite [Callee] type=peer host=voip1.other-domain.net deny=0.0.0.0/0.0.0.0 permit=yyy.yyy.yyy.yyy context=myOtherAccess disallow=all allow=all Now when I call from 115@caller to any number at callee side I'm rejected with Sending to xxx.xxx.xxx.xxx:5060 (no NAT) Using INVITE request as basis request - 281799ed7524c46966bcf303371ed...@xxx.xxx.xxx.xxx Found peer '115' for '115' from xxx.xxx.xxx.xxx:5060 --- Reliably Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060 --- SIP/2.0 401 Unauthorized This is, Asterisk try to authenticate on URI SIP user before from peer definition. If I change type from friend to peer it worked (I need the friend for this extension) Does someone has an idea on how to solve this problem? Thanks for any hint -- Daniel -- _ -- 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] Does Asterisk Support SIP Video Call ?
Le 16/11/2011 10:23, Faraj Khasib a écrit : Hi all, I tried making a video SIP call using Asterisk But it didnt workonly voice call works? Hi Faraj, Asterisk support H261, H263, H263+ and H264. Video calls are working since at least 1.4 version. You have to activate it by setting videosupport=yes in sip.conf -- Daniel -- _ -- 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 complete 2.5.0.1 - dahdi_dummy not compiled
Hi all, I have a question: we have few customers asterisk servers runing 1.4 1.6 or 1.8 asterisk version under Debian Lenny or Squeeze. No one of this computer has telephony card, so we use dahdi_dummy for timing. Asterisk and dahdi always compiled ourself (*) Last week we face quality problem on 3 of those servers and discovered that timing was bad and looking further, dahdi_dummy not compiled = not loaded! One server is stock Lenny 2.6.26, the other Lenny backport 2.6.32 and the last stock Squeeze 2.6.32, all those running asterisk 1.6.2.20/dahdi-complete 2.5.0.1 We checked with other installations who have dahdi_dummy loaded (asterisk 1.4 1.6 or 1.8 (*) this last with packages from asterisk.org) and timing is OK on those machines. How can we get dahdi_dummy compiled on those machines? Thanks for any hint -- Daniel -- _ -- 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 complete 2.5.0.1 - dahdi_dummy not compiled
Le 07/11/2011 10:19, Tzafrir Cohen a écrit : How can we get dahdi_dummy compiled on those machines? You no longer need to. Merely loading the module dahdi provides timing and pseudo channels for conferences if no DAHDI hardware is available. Well: output of 1.6.20 without dahdi_dummy Debian Squeeze (Bad) --- Results after 124 passes --- Best: 100.000 -- Worst: 99.604 -- Average: 99.882464, Difference: 100.001599 dh@pabx2:/etc/dahdi$ sudo lsmod|grep dahdi dahdi 171134 0 crc_ccitt 1323 1 dahdi output of 1.6.20 without dahdi_dummy Debian Lenny Backport (Bad) --- Results after 269 passes --- Best: 100.000 -- Worst: 99.607 -- Average: 99.953130, Difference: 99.999378 dh@kumquat:~$ sudo lsmod|grep dahdi dahdi 171150 26 crc_ccitt 1323 1 dahdi output of 1.6.20 with dahdi_dummy Debian Lenny (Good) --- Results after 184 passes --- Best: 100.000 -- Worst: 99.979 -- Average: 99.997323, Difference: 99.997337 dh@asterix:~$ lsmod|grep dahdi dahdi_dummy 8080 0 dahdi_transcode11912 1 wctc4xxp dahdi_voicebus 40768 2 wctdm24xxp,wcte12xp dahdi 200912 18 dahdi_dummy,xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp crc_ccitt 6528 1 dahdi output of 1.8.7-1 with dahdi_dummy from Ubuntu server package asterisk.org (Good) --- Results after 84 passes --- Best: 99.998 -- Worst: 99.993 -- Average: 99.996689, Difference: 99.996689 dh@bescomx:/var/log/asterisk$ sudo lsmod | grep dahdi dahdi_transcode 6836 0 dahdi_dummy 2760 0 dahdi 210885 2 dahdi_transcode,dahdi_dummy crc_ccitt 1675 1 dahdi Our problem is that on servers without dahdi_dummy (the 2 first) we face problem with cutted calls or bad audio (words are cutted or one syllabe of three). We are using SIP and ulaw/alaw codec. Face the same behavior with g722. Problem appears on phones (SNOM) connected directly to the servers and not to users using their own asterisk server connected to those two servers. -- Daniel -- _ -- 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] Asterisk 1.8.7 and client outside network
Hello everybody, sorry for delay Le 16/10/2011 16:51, Tarek Sawah a écrit : One more thing can you post your peer's configs as you have it in the config file? It's below, at the end of the original message. Tried as well type=peer with no luck Details: [snom320](!) type=peer host=dynamic context=default nat=no canreinvite=no qualify=yes dtmfmode=rfc2833 language=fr allow=all call-limit=2 busy-limit=2 language=fr mailbox=100 vmexten=090 [ulaw-phone](!) ; and another one for ulaw-only disallow=all allow=ulaw allow=alaw [callgroup1](!) callgroup=1 pickupgroup=1 and can you register with the same user from within the lan? Yes. And as I told, on 1.6.20 there is no problem. On the lan they are 5 SNOM300, 4 Siemens C610IP and 1 SNOM320 (not the same I tried to connect) and they are connecting well. Thanks for your help Date: Sun, 16 Oct 2011 12:33:27 +0200 From: ad...@tootai.net To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8.7 and client outside network Hi Tarek Le 15/10/2011 20:28, Tarek Sawah a écrit : Hello Daniel First question, do you have a firewall application or hardware installed on the network? The Asterisk server is also the firewall/router, iptables running on it. Second do you have some software similar to fail2ban? Yes, but I put the domain IP in ignoreip list. I checked fail2ban iptables rules, no trace of this IP Third check your IPTABLES if you can post the output of iptables-save would be good. if you can replace the localnet=Asterisk server external IP/32 with externip=Asterisk server external IP/32 I didn't send this info but externalip is setted to Asterisk server external IP/32 then we will be able to check your problem? This setup is working on tens of customers servers (1.2, 1.4 and 1.6), but this is the first one running 1.8 version. The same phone connect perfectly to our 1.6 server in the same conditions, so it's seems something related to 1.8 version. What I don't understand is that (violating IP ) should display the IP but in my case it's blank (or empty). Should domain contain as well the port despite the fact that we have insecure=port,invite? Thanks for your help Daniel Date: Sat, 15 Oct 2011 19:08:10 +0200 From: ad...@tootai.net To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8.7 and client outside network Hi, no clue on this? I found a thread in march from Faisal Hanif having the same problem but no one of the proposed ideas where working (reverse permit/deny, tried with only permit=0.0.0.0/0.0.0.0, aso), no luck :-) I don't now if it's solved for him. If someone had a solution on this, would be great to share ;-) Regards -- Daniel Le 07/10/2011 15:01, Administrator TOOTAI a écrit : Hi, my asterisk 1.8.7 is working well with phones (SNOM, Gigaset 620 and GrandStream) connected from the lan I now want to connect a snom320 from outside but it failed, having always [Oct 7 14:48:04] ERROR[3870]: netsock2.c:94 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported [Oct 7 14:48:04] WARNING[3870]: chan_sip.c:13597 parse_register_contact: Domain 'XX.XXX.XXX.XX:2048' disallowed by contact ACL (violating IP ) [Oct 7 14:48:04] WARNING[3870]: chan_sip.c:14306 register_verify: Registration denied because of contact ACL doesn't matter if I connect through a VPN or to the public IP using STUN. My sip.conf: localnet=172.24.0.0/12 localnet=169.254.0.0/255.255.0.0 ; Zero conf local network localnet=Asterisk server external IP/32 autodomain=yes ;allowexternaldomains=yes domain=172.24.30.250 ;Asterisk Server IP domain=Public Hostname domain=Another Public Hostname [309](snom320,ulaw-phone,callgroup1) type=friend insecure=port,invite secret=VoIP2auDIo contactdeny=0.0.0.0/0.0.0.0 contactpermit=XX.XXX.XXX.XX/32 ; External IP from phone, same as disallowed by contact ACL deny=0.0.0.0/0.0.0.0 permit=XX.XXX.XXX.XX/32 nat=yes Any clue? Why violating IP is empty? -- _ -- 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] Asterisk 1.8.7 and client outside network
Hi Tarek Le 15/10/2011 20:28, Tarek Sawah a écrit : Hello Daniel First question, do you have a firewall application or hardware installed on the network? The Asterisk server is also the firewall/router, iptables running on it. Second do you have some software similar to fail2ban? Yes, but I put the domain IP in ignoreip list. I checked fail2ban iptables rules, no trace of this IP Third check your IPTABLES if you can post the output of iptables-save would be good. if you can replace the localnet=Asterisk server external IP/32 with externip=Asterisk server external IP/32 I didn't send this info but externalip is setted to Asterisk server external IP/32 then we will be able to check your problem? This setup is working on tens of customers servers (1.2, 1.4 and 1.6), but this is the first one running 1.8 version. The same phone connect perfectly to our 1.6 server in the same conditions, so it's seems something related to 1.8 version. What I don't understand is that (violating IP ) should display the IP but in my case it's blank (or empty). Should domain contain as well the port despite the fact that we have insecure=port,invite? Thanks for your help Daniel Date: Sat, 15 Oct 2011 19:08:10 +0200 From: ad...@tootai.net To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8.7 and client outside network Hi, no clue on this? I found a thread in march from Faisal Hanif having the same problem but no one of the proposed ideas where working (reverse permit/deny, tried with only permit=0.0.0.0/0.0.0.0, aso), no luck :-) I don't now if it's solved for him. If someone had a solution on this, would be great to share ;-) Regards -- Daniel Le 07/10/2011 15:01, Administrator TOOTAI a écrit : Hi, my asterisk 1.8.7 is working well with phones (SNOM, Gigaset 620 and GrandStream) connected from the lan I now want to connect a snom320 from outside but it failed, having always [Oct 7 14:48:04] ERROR[3870]: netsock2.c:94 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported [Oct 7 14:48:04] WARNING[3870]: chan_sip.c:13597 parse_register_contact: Domain 'XX.XXX.XXX.XX:2048' disallowed by contact ACL (violating IP ) [Oct 7 14:48:04] WARNING[3870]: chan_sip.c:14306 register_verify: Registration denied because of contact ACL doesn't matter if I connect through a VPN or to the public IP using STUN. My sip.conf: localnet=172.24.0.0/12 localnet=169.254.0.0/255.255.0.0 ; Zero conf local network localnet=Asterisk server external IP/32 autodomain=yes ;allowexternaldomains=yes domain=172.24.30.250 ;Asterisk Server IP domain=Public Hostname domain=Another Public Hostname [309](snom320,ulaw-phone,callgroup1) type=friend insecure=port,invite secret=VoIP2auDIo contactdeny=0.0.0.0/0.0.0.0 contactpermit=XX.XXX.XXX.XX/32 ; External IP from phone, same as disallowed by contact ACL deny=0.0.0.0/0.0.0.0 permit=XX.XXX.XXX.XX/32 nat=yes Any clue? Why violating IP is empty? -- _ -- 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] Asterisk 1.8.7 and client outside network
Hi, no clue on this? I found a thread in march from Faisal Hanif having the same problem but no one of the proposed ideas where working (reverse permit/deny, tried with only permit=0.0.0.0/0.0.0.0, aso), no luck :-) I don't now if it's solved for him. If someone had a solution on this, would be great to share ;-) Regards -- Daniel Le 07/10/2011 15:01, Administrator TOOTAI a écrit : Hi, my asterisk 1.8.7 is working well with phones (SNOM, Gigaset 620 and GrandStream) connected from the lan I now want to connect a snom320 from outside but it failed, having always [Oct 7 14:48:04] ERROR[3870]: netsock2.c:94 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported [Oct 7 14:48:04] WARNING[3870]: chan_sip.c:13597 parse_register_contact: Domain 'XX.XXX.XXX.XX:2048' disallowed by contact ACL (violating IP ) [Oct 7 14:48:04] WARNING[3870]: chan_sip.c:14306 register_verify: Registration denied because of contact ACL doesn't matter if I connect through a VPN or to the public IP using STUN. My sip.conf: localnet=172.24.0.0/12 localnet=169.254.0.0/255.255.0.0 ; Zero conf local network localnet=Asterisk server external IP/32 autodomain=yes ;allowexternaldomains=yes domain=172.24.30.250 ;Asterisk Server IP domain=Public Hostname domain=Another Public Hostname [309](snom320,ulaw-phone,callgroup1) type=friend insecure=port,invite secret=VoIP2auDIo contactdeny=0.0.0.0/0.0.0.0 contactpermit=XX.XXX.XXX.XX/32 ; External IP from phone, same as disallowed by contact ACL deny=0.0.0.0/0.0.0.0 permit=XX.XXX.XXX.XX/32 nat=yes Any clue? Why violating IP is empty? Thanks for your help -- _ -- 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 1.8.7 and VoiceMailMain
Hi, We can't read the messages in our mailbox always getting -- SIP/tootaiAUDIO-0001 Playing '/var/spool/asterisk/voicemail/default/100/Old/msg0002.slin' (language 'fr') [Oct 11 13:24:50] WARNING[26778]: app_voicemail.c:7802 play_message: Playback of message /var/spool/asterisk/voicemail/default/100/Old/msg0002 failed As you see Asterisk try to read messages in slin format! Recorded messages that are present: root@bescomx:/var/spool/asterisk/voicemail/default/100/INBOX# ls -al total 944 drwxrwx--- 2 asterisk asterisk 4096 2011-10-11 13:24 . drwxrwx--- 6 asterisk asterisk 4096 2011-10-10 08:02 .. -rw-rw 1 asterisk asterisk 0 2011-10-08 07:20 msg.gsm -rw-rw-rw- 1 asterisk asterisk282 2011-10-08 07:20 msg.txt -rw-rw 1 asterisk asterisk 44 2011-10-08 07:20 msg.wav -rw-rw 1 asterisk asterisk 60 2011-10-08 07:20 msg.WAV -rw-rw 1 asterisk asterisk 7392 2011-10-10 19:12 msg0001.gsm -rw-rw-rw- 1 asterisk asterisk282 2011-10-10 19:12 msg0001.txt -rw-rw 1 asterisk asterisk 71724 2011-10-10 19:12 msg0001.wav -rw-rw 1 asterisk asterisk 7340 2011-10-10 19:12 msg0001.WAV -rw-rw 1 asterisk asterisk 14388 2011-10-11 05:32 msg0002.gsm -rw-rw-rw- 1 asterisk asterisk280 2011-10-11 05:32 msg0002.txt -rw-rw 1 asterisk asterisk 139564 2011-10-11 05:32 msg0002.wav -rw-rw 1 asterisk asterisk 14230 2011-10-11 05:32 msg0002.WAV -rw-rw 1 asterisk asterisk 20691 2011-10-11 06:24 msg0003.gsm -rw-rw-rw- 1 asterisk asterisk282 2011-10-11 06:24 msg0003.txt -rw-rw 1 asterisk asterisk 200684 2011-10-11 06:24 msg0003.wav -rw-rw 1 asterisk asterisk 20405 2011-10-11 06:24 msg0003.WAV -rw-rw 1 asterisk asterisk 2838 2011-10-11 06:30 msg0004.gsm -rw-rw-rw- 1 asterisk asterisk282 2011-10-11 06:30 msg0004.txt -rw-rw 1 asterisk asterisk 27564 2011-10-11 06:30 msg0004.wav -rw-rw 1 asterisk asterisk 2855 2011-10-11 06:30 msg0004.WAV -rw-rw 1 asterisk asterisk 14586 2011-10-11 07:08 msg0005.gsm -rw-rw-rw- 1 asterisk asterisk282 2011-10-11 07:08 msg0005.txt -rw-rw 1 asterisk asterisk 141484 2011-10-11 07:08 msg0005.wav -rw-rw 1 asterisk asterisk 14425 2011-10-11 07:08 msg0005.WAV -rw-rw 1 asterisk asterisk 1254 2011-10-11 08:00 msg0006.gsm -rw-rw-rw- 1 asterisk asterisk282 2011-10-11 08:00 msg0006.txt -rw-rw 1 asterisk asterisk 12204 2011-10-11 08:00 msg0006.wav -rw-rw 1 asterisk asterisk 1295 2011-10-11 08:00 msg0006.WAV -rw-rw 1 asterisk asterisk 7788 2011-10-11 08:00 msg0007.gsm -rw-rw-rw- 1 asterisk asterisk282 2011-10-11 08:00 msg0007.txt -rw-rw 1 asterisk asterisk 75564 2011-10-11 08:00 msg0007.wav -rw-rw 1 asterisk asterisk 7730 2011-10-11 08:00 msg0007.WAV -rw-rw 1 asterisk asterisk 5742 2011-10-11 08:06 msg0008.gsm -rw-rw-rw- 1 asterisk asterisk282 2011-10-11 08:06 msg0008.txt -rw-rw 1 asterisk asterisk 55724 2011-10-11 08:06 msg0008.wav -rw-rw 1 asterisk asterisk 5715 2011-10-11 08:06 msg0008.WAV Codec negotiation: Capabilities: us - 0x80030c7f (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719), peer - audio=0xc (ulaw|alaw)/video=0x38 (h263|h263p|h264)/text=0x0 (nothing), combined - 0x38000c (ulaw|alaw|h263|h263p|h264) In asterisk.conf we even activate transcode_via_sln = yes ;Build transcode paths via SLINEAR,instead of directly. Why is Asterisk trying to read messages in slin format? Thanks for any hint. -- Daniel -- _ -- 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] Asterisk 1.8.7 and ReceiveFAX
Le 07/10/2011 16:32, Kristijan Vrban a écrit : remove the c argument Done but now I have [Oct 8 19:20:20] WARNING[8771]: res_fax.c:1508 receivefax_t38_init: channel 'SIP/tootaiAUDIO-00ea' refused to negotiate T.38 [Oct 8 19:20:20] WARNING[8771]: res_fax.c:1529 receivefax_t38_init: Audio FAX not allowed on channel 'SIP/tootaiAUDIO-00ea' and T.38 negotiation failed; aborting. [Oct 8 19:20:20] ERROR[8771]: res_fax.c:1734 receivefax_exec: error initializing channel 'SIP/tootaiAUDIO-00ea' in T.38 mode How can I allow Audio FAX? I saw a discussion on asterisk-devel from january 2010 about new spandsp where Kevin P. Fleming told you to do an core show application ReceiveFAX to find out how to enable this feature. I'm perhaps a little bit stupid but can't find any usable information while using this command :-( Thanks for your help -- Daniel 2011/10/7 Administrator TOOTAIad...@tootai.net: Hi, I setup my first stock 1.8.7 asterisk (Ubuntu LTS 10.04 packages taken from deb http://packages.asterisk.org/deb lucid main) including dahdi from this same repository. No FFA involved. On incoming calls (only SIP, no telephony card), fax detection is working but reception failed with -- Executing [fax@from-TOOTAiAudio:19] ReceiveFAX(SIP/tootaiAUDIO-0564, /tmp/1317991071.1614.tiff,c) in new stack [Oct 7 14:37:52] WARNING[6961]: res_fax.c:1651 receivefax_exec: ReceiveFAX does not support polling == Spawn extension (from-TOOTAiAudio, fax, 19) exited non-zero on 'SIP/tootaiAUDIO-0564' What can be the problem? Thanks for any hint. -- _ -- 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] Asterisk 1.8.7 and ReceiveFAX
Le 08/10/2011 23:48, Bryant Zimmerman a écrit : The f/F option for ReceiveFAX is not in the 1.8.x builds. It was a patch for 1.8.x but it is in the 10 builds Well, I tried and it is working in 1.8.7 version, so command 'core show application ReceiveFAX' doesn't reflect the real application options, only shows c option which is not present in the link sended by Larry. Well ... FYI, I got this error -- Channel 'SIP/tootaiAUDIO-00ee' receiving FAX '/tmp/1318111488.262.tiff' [Oct 9 00:04:53] WARNING[9039]: res_fax.c:1508 receivefax_t38_init: channel 'SIP/tootaiAUDIO-00ee' refused to negotiate T.38 [Oct 9 00:05:05] WARNING[9039]: res_fax_spandsp.c:368 spandsp_log: WARNING T.30 ECM carrier not found -- Auto fallthrough, channel 'SIP/tootaiAUDIO-00ee' status is 'UNKNOWN' -- Executing [h@from-TOOTAiAudio:1] NoOp(SIP/tootaiAUDIO-00ee, Hangup Cause: 16) in new stack -- Executing [h@from-TOOTAiAudio:2] NoOp(SIP/tootaiAUDIO-00ee, Dial status : ) in new stack but the fax was received. Thanks Larry for the tip. *From*: Larry Moore lmo...@starwon.com.au *Sent*: Saturday, October 08, 2011 5:32 PM *To*: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] Asterisk 1.8.7 and ReceiveFAX On 9/10/2011 1:29 AM, Administrator TOOTAI wrote: Le 07/10/2011 16:32, Kristijan Vrban a écrit : remove the c argument Done but now I have [Oct 8 19:20:20] WARNING[8771]: res_fax.c:1508 receivefax_t38_init: channel 'SIP/tootaiAUDIO-00ea' refused to negotiate T.38 [Oct 8 19:20:20] WARNING[8771]: res_fax.c:1529 receivefax_t38_init: Audio FAX not allowed on channel 'SIP/tootaiAUDIO-00ea' and T.38 negotiation failed; aborting. [Oct 8 19:20:20] ERROR[8771]: res_fax.c:1734 receivefax_exec: error initializing channel 'SIP/tootaiAUDIO-00ea' in T.38 mode How can I allow Audio FAX? I saw a discussion on asterisk-devel from january 2010 about new spandsp where Kevin P. Fleming told you to do an core show application ReceiveFAX to find out how to enable this feature. I'm perhaps a little bit stupid but can't find any usable information while using this command :-( The Fallback option to T.30 is 'f'. ReceiveFAX(filename,f) See https://wiki.asterisk.org/wiki/display/AST/Application_ReceiveFAX+%28res_fax%29 -- Daniel -- _ -- 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 1.8.7 and ReceiveFAX
Hi, I setup my first stock 1.8.7 asterisk (Ubuntu LTS 10.04 packages taken from deb http://packages.asterisk.org/deb lucid main) including dahdi from this same repository. No FFA involved. On incoming calls (only SIP, no telephony card), fax detection is working but reception failed with -- Executing [fax@from-TOOTAiAudio:19] ReceiveFAX(SIP/tootaiAUDIO-0564, /tmp/1317991071.1614.tiff,c) in new stack [Oct 7 14:37:52] WARNING[6961]: res_fax.c:1651 receivefax_exec: ReceiveFAX does not support polling == Spawn extension (from-TOOTAiAudio, fax, 19) exited non-zero on 'SIP/tootaiAUDIO-0564' What can be the problem? Thanks for any hint. -- Daniel -- _ -- 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 1.8.7 and client outside network
Hi, my asterisk 1.8.7 is working well with phones (SNOM, Gigaset 620 and GrandStream) connected from the lan I now want to connect a snom320 from outside but it failed, having always [Oct 7 14:48:04] ERROR[3870]: netsock2.c:94 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported [Oct 7 14:48:04] WARNING[3870]: chan_sip.c:13597 parse_register_contact: Domain 'XX.XXX.XXX.XX:2048' disallowed by contact ACL (violating IP ) [Oct 7 14:48:04] WARNING[3870]: chan_sip.c:14306 register_verify: Registration denied because of contact ACL doesn't matter if I connect through a VPN or to the public IP using STUN. My sip.conf: localnet=172.24.0.0/12 localnet=169.254.0.0/255.255.0.0 ; Zero conf local network localnet=Asterisk server external IP/32 autodomain=yes ;allowexternaldomains=yes domain=172.24.30.250 ;Asterisk Server IP domain=Public Hostname domain=Another Public Hostname [309](snom320,ulaw-phone,callgroup1) type=friend insecure=port,invite secret=VoIP2auDIo contactdeny=0.0.0.0/0.0.0.0 contactpermit=XX.XXX.XXX.XX/32 ; External IP from phone, same as disallowed by contact ACL deny=0.0.0.0/0.0.0.0 permit=XX.XXX.XXX.XX/32 nat=yes Any clue? Why violating IP is empty? Thanks for your help -- Daniel -- _ -- 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] Core show translation 4000ms
Hi list, we have 2 asterisk boxes in VM (kvm) on 2 different Dell servers, one is Lenny kernel 2.6.26 asterisk 1.6.2.20, the second CentOS 2.6.18 asterisk 1.4.36 (Elastix). Both 64bits, no hardware involved, dahdi on both machines for meetme timing. Doing core show translation give on the Lenny server Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16 g723 - - - -- - - - - - - - - - - - gsm - - 2 2 4001 2 1 2 - - - 4001 4002 - - 4003 ulaw - 4001 - 1 4001 2 1 2 - - - 4001 4002 - - 4003 alaw - 4001 1 - 4001 2 1 2 - - - 4001 4002 - - 4003 [...] and on the CentOS one g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 g723- ---- -- -- - --- gsm- -222 21 3- 6 -22 ulaw- 2-12 21 3- 6 -22 alaw- 21-2 21 3- 6 -22 [...] Why do we have such latency on the Lenny machine for the codec translation? Is this due to a kernel parameter? Thanks for any hint -- Daniel -- _ -- 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] Core show translation 4000ms
Le 30/09/2011 14:05, Kevin P. Fleming a écrit : On 09/30/2011 03:56 AM, Administrator TOOTAI wrote: Hi list, we have 2 asterisk boxes in VM (kvm) on 2 different Dell servers, one is Lenny kernel 2.6.26 asterisk 1.6.2.20, the second CentOS 2.6.18 asterisk 1.4.36 (Elastix). Both 64bits, no hardware involved, dahdi on both machines for meetme timing. Doing core show translation give on the Lenny server Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16 g723 - - - - - - - - - - - - - - - - gsm - - 2 2 4001 2 1 2 - - - 4001 4002 - - 4003 ulaw - 4001 - 1 4001 2 1 2 - - - 4001 4002 - - 4003 alaw - 4001 1 - 4001 2 1 2 - - - 4001 4002 - - 4003 [...] and on the CentOS one g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 g723 - - - - - - - - - - - - - gsm - - 2 2 2 2 1 3 - 6 - 2 2 ulaw - 2 - 1 2 2 1 3 - 6 - 2 2 alaw - 2 1 - 2 2 1 3 - 6 - 2 2 [...] Why do we have such latency on the Lenny machine for the codec translation? Is this due to a kernel parameter? Because you didn't read the output. It clearly says (in microseconds) in the 1.6.x output. Well, I surely ask the wrong way, sorry: ms or us, 4001 from ulaw to gsm and 2 the other way, still a huge difference. The output from centos shows similar value in both directions. -- Daniel -- _ -- 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] Core show translation 4000ms
Le 30/09/2011 16:59, Eric Wieling a écrit : I always use the recalc option to show translations, it seems to provide much more accurate numbers. Example: core show translation recalc 20 Lenny kernel, new values, still 1000 microseconds between both directions Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16 g723 - - - -- - - - - - - - - - - - gsm - - 601 601 3800 800 600 2000 - - - 3800 1200 - - 2000 ulaw - 1601 - 1 3201 201 1 1401 - - - 3201 601 - - 1401 alaw - 1601 1 - 3201 201 1 1401 - - - 3201 601 - - 1401 CentOS, no changes g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 g723- ---- -- -- - --- gsm- -222 21 5- 7 -22 ulaw- 2-12 21 5- 7 -22 alaw- 21-2 21 5- 7 -22 I ran the same command on an Squeeze 2.6.32 kernel running 1.8.7 asterisk: values are neer those from CentOS asterisk 1.4 version -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tony Mountifield Sent: Friday, September 30, 2011 10:54 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Core show translation 4000ms Maybe, but that still doesn't explain why there is a factor of 2000 between some conversions and others. And 4001, 4002 and 4003 are remarkably like a big round number plus a tiny offset! I would agree with the OP that the values shown look suspicious and would bear some investigating... [...] -- Daniel -- _ -- 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] Core show translation 4000ms
Le 30/09/2011 17:02, Jason Parker a écrit : On 09/30/2011 09:53 AM, Tony Mountifield wrote: In article4e85d19f.4090...@digium.com, Kevin P. Flemingkpflem...@digium.com wrote: This is why the output was changed to microseconds from milliseconds; in the older version, the lowest number that should be shown was 1 millisecond, even if the actual amount of time consumed was 10 microseconds (or less). The 1 numbers in the output from the older could easily have been 0.02, which would be closer to the output from the new version. Maybe, but that still doesn't explain why there is a factor of 2000 between some conversions and others. And 4001, 4002 and 4003 are remarkably like a big round number plus a tiny offset! I would agree with the OP that the values shown look suspicious and would bear some investigating... I believe the way it gets calculated was also changed a bit. You'll commonly see numbers that are near multiples of 1000. If I'm not mistaken these are the duration of a context switch (or several context switches), which means that with this output, you can guess that his kernel is probably compiled with CONFIG_HZ_250. As Tony pointed out, it's the factor between both translation directions which push me to ask. I can leave with microseconds and understand the why, but values should not have a so big interval. -- Daniel -- _ -- 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] Read command - input correction not taken in account
Hi all, using asterisk 1.4 or 1.6, I face a problem with the read command. I call my asterisk box which ask me to enter the number I wish to call. Problem is that if I make a mistake in the number and correct it on the phone keyboard (smartphone under android, the same with nokias series E), asterisk already took the digit and just append the next one insteed of replacing the previous one as shown on the phone display. Is there a way of getting this working as expected with the read command? Another solution? Thanks for any hint -- Daniel -- _ -- 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] Can't get video on one server of 4
Hi, we have 4 asterisk, versions are 1.4.35 1.4.36 1.6.2.18 and 1.4.42 One GrandStream GXV3000 is used for the tests. He is registered to asterisk 1.6.2.18 asa well as 1.4.35. Calling echo test is OK on both servers, get audio and video. Calling echo test from asterisk 1.4.36 bye a SIP trunk from both others servers is also working well. What fail, is video on echo test from asterisk 1.4.42 using SIP trunks: we have audio but no videobeside the fact that video codec are negociated as shown below. All servers are on public IP. Here is a debug from a call from server running 1.4.35 asterisk to the 1.4.42 one: - [2011-07-05 16:08:14] VERBOSE[11535] logger.c: --- (14 headers 18 lines) --- [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Sending to XXX.XXX.XXX.XXX : 5060 (no NAT) [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Using INVITE request as basis request - 78938c042d374b341c4f1b60071d3...@xxx.xxx.xxx.xxx [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found peer 'mypeer' [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found RTP audio format 0 [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found RTP audio format 3 [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found RTP audio format 101 [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found audio description format PCMU for ID 0 [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found audio description format GSM for ID 3 [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found audio description format telephone-event for ID 101 [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found RTP video format 34 [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found RTP video format 103 [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found RTP video format 99 [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found video description format H263 for ID 34 [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found video description format h263-1998 for ID 103 [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Found video description format H264 for ID 99 [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Capabilities: us - 0x3c0002 (gsm|h261|h263|h263p|h264), peer - audio=0x380006 (gsm|ulaw|h263|h263p|h264)/video=0x38 (h263|h263p|h264), combined - 0x380002 (gsm|h263|h263p|h264) [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Peer audio RTP is at port XXX.XXX.XXX.XXX:40428 [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Peer video RTP is at port XXX.XXX.XXX.XXX:44636 [2011-07-05 16:08:14] VERBOSE[11535] logger.c: Looking for 3800 in acces_groupe (domain mydomain.com) [2011-07-05 16:08:14] VERBOSE[11535] logger.c: list_route: hop: sip:3...@xxx.xxx.xxx.xxx [2011-07-05 16:08:14] VERBOSE[11535] logger.c: --- Transmitting (NAT) to XXX.XXX.XXX.XXX:5060 --- All trunks. are setted from the same manier: [trunk] ; type=peer deny=0.0.0.0/0.0.0.0 permit=XXX.XXX.XXX.XXX host=host.domain.com context=from-trunk disallow=all allow=all A sip show peer mypeer show that video is on. What can be the problem, I start loose my hairs! Thanks for any hint. -- Daniel -- _ -- 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] Inbound SMS
Le 22/06/2011 01:10, ERIC HERRON a écrit : I know Asterisk 1.8 can send out texts via SMS() Can I send Asterisk a text via a DID and it do something? [...] You can receive SMSs using smsq (at least in 1.4) But be aware that most of mobile carriers (eg France) send SMSs to landlines number as voicemail message :-( -- Daniel -- _ -- 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] Asterisk 1.4.41 - Warning and Notice about contact info and stale nonce
Hi, Nobody on this? Le 16/05/2011 23:35, Administrator TOOTAI a écrit : Le 16/05/2011 18:27, Jose P. Espinal a écrit : Administrator TOOTAI wrote: Of course it's 1.4.41. And the result is that devices doesn't register anymore. Thanks for any hint. If you are installing from source, check out if some modules did not load properly due to undefined symbols. # asterisk -gvvc | tee output.txt CLI stop gracefully Then review that output.txt file. Don't think that the problem is here: the devices are working well with previous version of asterisk on the same server. Also, other devices from other manufacturer are still working ok. Question is why auth is OK but registration failed? On 1.4.40 we juste had to change the device local port (eg from 5061 to 5062) and registration was OK. On 1.4.41 this trick is no more working. And stale nonce should have an end of life in our mind, but doesn't. Thanks for your tip. -- Daniel -- _ -- 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] Asterisk 1.4.41 - Warning and Notice about contact info and stale nonce
Of course it's 1.4.41. And the result is that devices doesn't register anymore. Thanks for any hint. Le 14/05/2011 17:37, Administrator TOOTAI a écrit : Hi list, We have devices since more then 4 years which where running well with Asterisk. But with latest version (1.38 or more) we face problem with those devices when they try to register. We got [2011-05-14 17:18:06] WARNING[28559]: chan_sip.c:9950 register_verify: Failed to parse contact info --- Transmitting (NAT) to XXX.XXX.XXX.XXX:5062 --- SIP/2.0 400 Bad Request Followed by [2011-05-14 17:19:06] NOTICE[28559]: chan_sip.c:9502 check_auth: Correct auth, but based on stale nonce received from 'sip:7...@yyy.yyy.yyy.yyy;user=phone;tag=63d2ba80bffb016f' Checking logs we found Contact: * in headers before the failed parse contact info. We checked in source chan_sip and saw the parse info reject with Error 400 after the auth is correct comment. We modified in sip.conf the type=peer in type=friend, same result. Could someone explain us what happends here? Thanks -- Daniel -- _ -- 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] Asterisk 1.4.41 - Warning and Notice about contact info and stale nonce
Le 16/05/2011 18:27, Jose P. Espinal a écrit : Administrator TOOTAI wrote: Of course it's 1.4.41. And the result is that devices doesn't register anymore. Thanks for any hint. If you are installing from source, check out if some modules did not load properly due to undefined symbols. # asterisk -gvvc | tee output.txt CLI stop gracefully Then review that output.txt file. Don't think that the problem is here: the devices are working well with previous version of asterisk on the same server. Also, other devices from other manufacturer are still working ok. Question is why auth is OK but registration failed? On 1.4.40 we juste had to change the device local port (eg from 5061 to 5062) and registration was OK. On 1.4.41 this trick is no more working. And stale nonce should have an end of life in our mind, but doesn't. Thanks for your tip. -- Daniel -- _ -- 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 1.41 - Warning and Notice about contact info and stale nonce
Hi list, We have devices since more then 4 years which where running well with Asterisk. But with latest version (1.38 or more) we face problem with those devices when they try to register. We got [2011-05-14 17:18:06] WARNING[28559]: chan_sip.c:9950 register_verify: Failed to parse contact info --- Transmitting (NAT) to XXX.XXX.XXX.XXX:5062 --- SIP/2.0 400 Bad Request Followed by [2011-05-14 17:19:06] NOTICE[28559]: chan_sip.c:9502 check_auth: Correct auth, but based on stale nonce received from 'sip:7...@yyy.yyy.yyy.yyy;user=phone;tag=63d2ba80bffb016f' Checking logs we found Contact: * in headers before the failed parse contact info. We checked in source chan_sip and saw the parse info reject with Error 400 after the auth is correct comment. We modified in sip.conf the type=peer in type=friend, same result. Could someone explain us what happends here? Thanks -- Daniel -- _ -- 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] Discussion: Are we ready to leave 1.4 behind?
Le 29/04/2011 00:42, Russell Bryant a écrit : - Original Message - Sure. Please follow the 2 next stories: - had a customer running 1.4.26 We upgraded to a new server and installed 1.4.39, last version at this time. Bang: voicemail doesn't work as it should, had to fallback to 1.4.26 Customer is still running this version. - have 1.4.41 and 1.6.16 which are no more able to use auth keys in iax since we update one server from 1.4 to 1.6 Now imagine that 1.4 stays at only security level. For first case we have 2 options: upgrading for security reasons to last version but then no more voicemail, or staying with 1.4.26. In the second case, upgrading both servers to test with 1.8. If it's still not working, it was time loose beside other problems. If there are obvious regressions in major functionality such as voicemail, I'm more than happy to still consider making fixes for those problems during the security maintenance period. It has to be pretty clear, though, and in this particular case, it is. Can you point to the bug number please? I want to make sure this voicemail problem is resolved as soon as possible. https://issues.asterisk.org/view.php?id=18998 for the voicemail https://issues.asterisk.org/view.php?id=18539 for the iax2 auth rsa -- Daniel -- _ -- 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] Discussion: Are we ready to leave 1.4 behind?
Le 28/04/2011 16:53, Russell Bryant a écrit : - Original Message - PS. Please don't start a discussion about 1.8 quality in this thread, that's a separate issue. I just want to know what you think about closing 1.4 support now. If you want to discuss 1.8 quality, start a new thread. Thanks. I don't think it's a separate issue at all. I would like to see discussion of exactly which issues are preventing users from using Asterisk 1.8. We're trying to shift focus to those issues and get them resolved as quickly and as efficiently as we can so that we can all move forward. Let's see it from another angle: we today are mainly using 1.4 and 1.6.2 In the last month with faced those regressions, first still not solved: https://issues.asterisk.org/view.php?id=18539 https://issues.asterisk.org/view.php?id=18998 Do you think we're ready to switch to 1.8 if 1.4/1.6 still have such behavior? As I told in previous answer, we started 1.4 in production very early and had lots of troubles, we don't want to face the same over activity with 1.8 Resources are limited. This I understand What is the best use of our time to help ensure the best future? Where do we want to see the project in the next 6 months to a year? A primary focus on further solidifying Asterisk 1.8 is what gets us there in my mind. Agree Asterisk 1.4 was released 4.5 years ago. It mostly just works, and I fully expect many to keep using it until they see a need to migrate. This process has been likened to when the community moved from Asterisk 1.2 to 1.4. Asterisk 1.8 has been much more stable out of the gate than 1.4, due to many things we have done over the years to increase quality, including: [...] Ok, so why not stay with asterisk 1.4 security *and* bug/regression fixes for few weeks/monthes till 1.8 reaches the level that the community accept to switch to 1.8 -- Daniel -- _ -- 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] Discussion: Are we ready to leave 1.4 behind?
Le 28/04/2011 21:47, Leif Madsen a écrit : On 11-04-28 12:04 PM, Administrator TOOTAI wrote: Ok, so why not stay with asterisk 1.4 security *and* bug/regression fixes for few weeks/monthes till 1.8 reaches the level that the community accept to switch to 1.8 What is the guide here? What is the level that the community accepts? Unfortunately that is a statement that is impossible to measure quantitatively. The answer will always be, We're not ready! Don't think so, analyze the answers to this discussion -thanks Ole ;-)-: till 1.8 is not at the feature level and stability of 1.4, people like me will not move to 1.8 Measure is easy :-) Having to focus on issues on both the 1.4 and 1.8 branches simultaneously distracts from the goal of making 1.8 stable (which in my several deployments recently, it seems to be). Again, I think that maintaining 1.4 on his today level is ok *if and only if* bugs/regression are taking in account, not only security. [...] With focus being directed to 1.8, the issues that may be blocking you from having a successful migration to, or deployment of, Asterisk 1.8 will get fixed that much sooner. In production you can't use something which will be fixed sooner. It has to work straight on, at least when you upgrade from a previous version. Customer doesn't care if the new version is more up to date and has new features if in the mean time they don't have features that they had before. If the community won't, or can't, step up to maintain a community based branch which has very few changes being made to it, then I'm not sure it is fair to expect Digium to do that. That's one point for you: community seems to say we want that 1.4 still lives but no one [want|doesn't have the knowledge] to participate on maintaining the community branch. -- Daniel -- _ -- 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] Discussion: Are we ready to leave 1.4 behind?
Le 28/04/2011 22:43, Leif Madsen a écrit : On 11-04-28 04:33 PM, Administrator TOOTAI wrote: Le 28/04/2011 21:47, Leif Madsen a écrit : On 11-04-28 12:04 PM, Administrator TOOTAI wrote: Ok, so why not stay with asterisk 1.4 security *and* bug/regression fixes for few weeks/monthes till 1.8 reaches the level that the community accept to switch to 1.8 What is the guide here? What is the level that the community accepts? Unfortunately that is a statement that is impossible to measure quantitatively. The answer will always be, We're not ready! Don't think so, analyze the answers to this discussion -thanks Ole ;-)-: till 1.8 is not at the feature level and stability of 1.4, people like me will not move to 1.8 Measure is easy :-) But that's what I don't get. No one is *forcing* you to move to 1.8 *right now*. The code base for 1.4 isn't going anywhere. Anyone is able to keep deploying 1.4 (or 1.2, or 1.0, or 0.9 for that matter) to their hearts content. Sure. Please follow the 2 next stories: - had a customer running 1.4.26 We upgraded to a new server and installed 1.4.39, last version at this time. Bang: voicemail doesn't work as it should, had to fallback to 1.4.26 Customer is still running this version. - have 1.4.41 and 1.6.16 which are no more able to use auth keys in iax since we update one server from 1.4 to 1.6 Now imagine that 1.4 stays at only security level. For first case we have 2 options: upgrading for security reasons to last version but then no more voicemail, or staying with 1.4.26. In the second case, upgrading both servers to test with 1.8. If it's still not working, it was time loose beside other problems. Yes, we have servers for testing, but really, who would think that such 2 problems araised with an 1.4 stable version? Same was few versions before (1.4.20~1.4.28 if I good remember) with attempted call transfer: was working on one version, stop to next one, worked again aso. Even in a test environment you can't simulate all setups. Hope that this both scenario gives you a new vision ;-) and why I tell that bugs and regressions should be taken in account at the same level as security. -- Daniel -- _ -- 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] Discussion: Are we ready to leave 1.4 behind?
Le 27/04/2011 21:34, Olle E. Johansson a écrit : Friends, We have a discussion on asterisk-dev about the maintenance of the 1.4 branch. According to the release plans, support for 1.4 was scheduled to close in April 2011 - basically now. After that, only security patches would be committed. This is already a delay from the original plan published by Russell Bryant. Unfortunately, I think this is way too early. My feeling and experience is that 1.8 is not ready for production in the environments I work in - large scale installations. Customers are not planning migration and all new installs are still 1.4. Tests we've been doing with 1.8 has failed within just a short time and so badly that customers has not paid me to spend any further time with 1.8. [...] Agree with you at 100%. 1.8 is not ready for production. I remember our switch from 1.2 to 1.4 very early and had huge problems (misdn and B410P just comes in my mind), had to work with trunk, aso. At 1.4.8 or so it started to be stable. We're now at 1.8.3 ... Also, latest 1.4 had some regressions (eg voicemailbox sequences), which means that we're not, at this time, sure that basic stuffs are working smoothly with 1.4.41 What happends if new regressions appears? My vote goes to stay with 1.4 and continue to stabilize it (not asking to include new stuff) till community declare that 1.8 is at the level of 1.4. -- Daniel -- _ -- 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] asterisk and fail2ban
Le 29/03/2011 19:34, Sherwood McGowan a écrit : On 3/29/2011 12:25 PM, Steve Edwards wrote: On Tue, 29 Mar 2011 12:10:59 -0500, Sherwood McGowan First thing I'd do is restrict the ip blocks your sip endpoints can register/call from in sip.conf (or your database's table for sip endpoints) On Tue, 29 Mar 2011, Gilles wrote: Thanks for the idea, but it's not possible, as the Asterisk must be accessible for road warriors and receive SIP calls from anyone. Really? How many callers are you expecting from North Korea, Libya, China, Iran, etc? Thanks Steve, you just emailed exactly what I was going to say... Remember guys, there's a LOT of IP blocks out there that are almost definitely not going to be somewhere you expect to receive SIP traffic from. Well, I can tell you that our servers in europe those days are mainly attacked by US IP ranges (remember last year the problem with amazon cloud). They now disappear here in europe but lots of other US networks quickly replace them :-( -- Daniel -- _ -- 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] Channel status with AMI originate calls
Hi, is there a way to know if originate call channel ended the call *before* connecting to context/extension/priority? DIALSTATUS is empty, HANGUPCAUSE is always 16, nothing in SIP Headers nor in AST_CONTROL_FRAME_[HANGUP|ANSWER] Asterisk is 1.6.2.16 Thanks for any hint -- Daniel -- _ -- 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] issue with some numbers
Le 14/02/2011 15:44, salaheddine elharit a écrit :
thanks for your response
i have tested with a regular phone and i get the same result
my question if there is any action to do in dial plan or
extenssion.conf in order to call this number becouse in dial plan i can
bloc a number to be call
exten = _OUT.,n,Set(match=${REGEX(^06XXX ${AH_PHONE_NUMBER})})
exten = _OUT.,n,GotoIf($[${match} = 1]?rien)
I think you're in France, your REGEX is wrong: France has 10 digits
(one X missing). Also, remember that since few monthes, French mobile
phones can also start with 07.
Regards
--
Daniel
--
_
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Re: [asterisk-users] Sound quality issue
Le 15/01/2011 20:38, Cédric Lemarchand a écrit : Hello, Hi [...] I am sure there are RTP packets losses somewhere, except RTP debug in the asterisk CLI, how can i determine where the problem come from ? [...] You don't tell which protocol (SIP, IAX, H323) nor which asterisk version. FYI Asterisk 1.6.2.15 in iax had audio quality problems, solved in 1.6.2.16. If you have the possibility, connect directly a phone to the server, eg Device - LAN (no MPLS) - Asterisk. Check also if a call to echo test has the same bad quality. -- Daniel -- _ -- 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] Log and forward calls to cellphone?
Le 04/01/2011 20:50, Sebastian a écrit : Hi, On 01/04/2011 03:24 PM, Administrator TOOTAI wrote: Le 04/01/2011 11:50, Gilles a écrit : [...] It looks like getting a 3G smartphone with SIP + OpenVPN + unlimited Internet plan would solve the issue. I Would avoid OpenVPN (tested an Android) as it drains quickly battery Any chance you could provide few more details please? Mainly which phone, what version of Android, and how many hours on standby when using OpenVPN. Also, which application were you running through OpenVPN and was it in constant use (the app). Hmmh, most of all those infos were given in the original message, see below ;-). HTC Hero rooted with Android 2.1 VillainRom9.0.0 Sip client is SipDroid (tested few others but never got them connecting to our Asterix). OpenVPN drains battery in less then 4 hours without calling. SipDroid is able to connect using 3G, I use it from time to time. How I use my mobile phone: . in the office, connected through WIFI with Asterisk server: can pass and receive calls, any technologie . out of the office: incoming calls to office numbers are routed to my mobile number after x seconds of no answer from the office phones. My mobile subscription include free calls to few landlines numbers 24h/24h 7d/7d: one of them is the office number. Calling this number give me an IVR from where I can enter the number I wish to call using our SIP routes. As I told, the best SIP client I had is Nokias one. Fully integrated, working out of the box. I am investigating using OpenVPN with Android - and I would find the above detail very useful. Many thanks, Sebastian [...] 2. what smartphone supports installing an SIP + OpenVPN clients? Without OpenVPN lots off, IPhone, Android, Nokia, Windows mobile, ... Best SIP client integrated with mobile are Nokias (E series for instance). I'm running HTC Hero (Android) with SipDroid. [...] -- Daniel -- _ -- 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] Log and forward calls to cellphone?
Le 03/01/2011 18:28, Gilles a écrit : On Mon, 03 Jan 2011 12:27:56 +0100, Administrator TOOTAI ad...@tootai.net wrote: As you are a Free Telecom customer, why not using your freephonie account to forward incoming calls to your mobile? Thanks for the tip, but experience shows that their SIP access sucks (not reliable, quality NOK). That's why I got a VOSP account. Don't know the meaning of VOSP but you can do it with any SIP/IAX/H323/... provider. -- Daniel -- _ -- 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] Log and forward calls to cellphone?
Le 04/01/2011 11:50, Gilles a écrit : [...] It looks like getting a 3G smartphone with SIP + OpenVPN + unlimited Internet plan would solve the issue. I Would avoid OpenVPN (tested an Android) as it drains quickly battery [...] 2. what smartphone supports installing an SIP + OpenVPN clients? Without OpenVPN lots off, IPhone, Android, Nokia, Windows mobile, ... Best SIP client integrated with mobile are Nokias (E series for instance). I'm running HTC Hero (Android) with SipDroid. [...] -- Daniel -- _ -- 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] Log and forward calls to cellphone?
Le 01/01/2011 18:32, Gilles a écrit :
On Wed, 29 Dec 2010 16:55:46 +0100, Administrator TOOTAI
ad...@tootai.net wrote:
I wouldn't be one of your friend: when I'm calling you I call a landline
but finally will be charged for a mobile call (imagine I have free calls
to landlines from my ISP). I give you an information: in France you
don't have the right to do this unless you have it precise *before*
redirection.
I checked with the VOSP: Apparently, it doesn't support getting an SIP
message to forward calls on the fly, and I pay for the forwarded leg
of the call (the caller will pay his part).
As you are a Free Telecom customer, why not using your freephonie
account to forward incoming calls to your mobile?
Something like in you POTS incoming context:
...
exten =
s,n,Dial(SIP/${Phone1}SIP/{MobilePhoneConnectedWithWIFI}IAX2/${SoftPhone},21,${DIAL_OPTIONS})
exten = s,n,Goto(s-${DIALSTATUS},1)
exten =
s-NOANSWER,1,Dial(SIP/freephonie/${MyMobileNumber},30,${DIAL_OPTIONS})
exten = s-NOANSWER,n,Hangup
exten = s-ANSWER,1,Hangup
exten =_s-.,1,Voicemail();other cases
--
Daniel
--
_
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Re: [asterisk-users] Asterisk 1.4.38 - unknown signalling bri_cpe
Le 28/12/2010 20:31, Kevin P. Fleming a écrit : [...] If you have a suggestion for a better place for this information to be made available, please let us know. [...] For instance in overview of Hx8: New with the release of the H8 cards is Digium's B400M four-port EuroISDN S/T module. The B400M sets a new standard for BRI connectivity in the Asterisk market with its support for software-selectable mode (NT or TE) and line termination. The B400M requires no jumpers for operation, regardless of mode or termination. This module is compatible with Asterisk starting from 1.6 version and above -- Daniel -- _ -- 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] Log and forward calls to cellphone?
Le 29/12/2010 12:16, Gilles a écrit : [...] In case a call comes in and I'm not home, I'd like Asterisk to log the call, and then send an SIP message to my VOSP so the call is forwarded to my cellphone and is thus charged to the caller, without Asterisk having to dial out to my cellphone through my VOSP at my expense and bridge the two calls. [...] I wouldn't be one of your friend: when I'm calling you I call a landline but finally will be charged for a mobile call (imagine I have free calls to landlines from my ISP). I give you an information: in France you don't have the right to do this unless you have it precise *before* redirection. Perhaps I misundersand you ... -- Daniel -- _ -- 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] Asterisk 1.4.38 - unknown signalling bri_cpe
Le 27/12/2010 20:09, Kevin P. Fleming a écrit : On 12/27/2010 12:37 PM, Administrator TOOTAI wrote: [...] d...@myphoneserver:/usr/src$ strings /usr/src/asterisk-1.4.38/channels/chan_dahdi.so | grep '^DAHDI Telephony' DAHDI Telephony w/PRI DAHDI Telephony Driver w/PRI Asterisk 1.4 has never had BRI support in chan_dahdi. Ouch :-( Are Digiums HB8 cards working with mISDN and Asterisk 1.4? -- Daniel -- _ -- 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] Asterisk 1.4.38 - unknown signalling bri_cpe
Le 28/12/2010 13:10, Kevin P. Fleming a écrit : On 12/28/2010 05:17 AM, Administrator TOOTAI wrote: Le 27/12/2010 20:09, Kevin P. Fleming a écrit : On 12/27/2010 12:37 PM, Administrator TOOTAI wrote: [...] d...@myphoneserver:/usr/src$ strings /usr/src/asterisk-1.4.38/channels/chan_dahdi.so | grep '^DAHDI Telephony' DAHDI Telephony w/PRI DAHDI Telephony Driver w/PRI Asterisk 1.4 has never had BRI support in chan_dahdi. Ouch :-( Are Digiums HB8 cards working with mISDN and Asterisk 1.4? Not to my knowledge; Digium has not produced an mISDN driver for the HX series cards, and I doubt anyone else has. You should modify your ADL_quickstart document on Digium store to precise the Asterisk version compatible with those cards (perhaps also in datasheet or somewhere else). At this time you have Asterisk-X.X-current.tar.gz but as 1.4, 1.6.2 and 1.8 are existing in current, I will not be the only one making this mistake. -- Daniel -- _ -- 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] One way crappy audio in iax call - Asterisk 1.6.2.15
Le 24/12/2010 16:47, Steve Davies a écrit : On 24 December 2010 14:40, Administrator TOOTAIad...@tootai.net wrote: Hi, We had 2 asterisk 1.4 connected together in iax, all was fine. One of them was upgraded (server and Asterisk) in 1.6.2.15, the other end is in 1.4.38 When calling to 1.4 to 1.6.2 -remember, it's iax- all is good. But calling from 1.6.2 to 1.4 give a bad audio to calling party (words are cutted, you can't understand the words). On callee party it's still good. We replace 1.6.2 version with 1.4.38 and everything is going back to normal, good audio on both side does'nt matter who call. I already opened another thread about problem with iax and Asterisk 1.6.2 (rsa auth not working anymore). Are there some known problems with iax and 1.6 version of Asterisk? Thanks for any hint Not 100% sure, but I think there was a fix for IAX audio in 1.6.2.16-rc1 - Perhaps try that? Done and it effectively seems to solve the problem. Thanks. -- Daniel -- _ -- 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 attack.. fail2ban not stopping attack
Le 27/12/2010 16:20, dave george a écrit :
[...]
[Definition]
#_daemon = asterisk
# Option: failregex
# Notes.: regex to match the password failures messages in the logfile. The
# host must be matched by a group named host. The tag HOST
can
# be used for standard IP/hostname matching and is only an alias
for
# (?:::f{4,6}:)?(?Phost\S+)
# Values: TEXT
#
failregex = NOTICE.* .*: Registration from '.*' failed for 'HOST' - Wrong
password
NOTICE.* .*: Registration from '.*' failed for 'HOST' - No
matching peer found
NOTICE.* .*: Registration from '.*' failed for 'HOST' -
Username/auth name mismatch
NOTICE.* .*: Registration from '.*' failed for 'HOST' - Device
does not match ACL
NOTICE.*HOST failed to authenticate as '.*'$
NOTICE.* .*: No registration for peer '.*' \(fromHOST\)
NOTICE.* .*: HostHOST failed MD5 authentication for '.*' (.*)
NOTICE.* .*: Failed to authenticate user .*@HOST.*
ignoreregex =
[...]
How looks your asterisk notice file?
---
Daniel
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[asterisk-users] Asterisk 1.4.38 - unknown signalling bri_cpe
Hi, we upgraded an Asterisk 1.4 with mISDN to 1.6 with chan_dahdi. Due to problems with iax channel posted earlier, we wanted to switch back to 1.4 version. Server has 2 HBA cards, everything is running fine with 1.6, bri_cpe is recognized and the 7 euroISDN channels are running well, ingoing and outgoing. Now we installed 1.4.38 version and no more ISDN. In logs we found this: [2010-12-24 14:50:38] VERBOSE[1773] logger.c: == Parsing '/etc/asterisk/chan_dahdi.conf': [2010-12-24 14:50:38] VERBOSE[1773] logger.c: Found [2010-12-24 14:50:38] ERROR[1773] chan_dahdi.c: Unknown signalling method 'bri_cpe' [2010-12-24 14:50:38] ERROR[1773] chan_dahdi.c: Signalling must be specified before any channels are. We think about a bug in libpri 1.4.11.4 so installed 1.4.11.5, same result. Dahdi linux and tools are 2.4.0 And yes, Asterisk is build with libpri ;-) d...@myphoneserver:/usr/src$ strings /usr/src/asterisk-1.4.38/channels/chan_dahdi.so | grep '^DAHDI Telephony' DAHDI Telephony w/PRI DAHDI Telephony Driver w/PRI Thanks for your help -- Daniel -- _ -- 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] One way crappy audio in iax call - Asterisk 1.6.2.15
Hi, We had 2 asterisk 1.4 connected together in iax, all was fine. One of them was upgraded (server and Asterisk) in 1.6.2.15, the other end is in 1.4.38 When calling to 1.4 to 1.6.2 -remember, it's iax- all is good. But calling from 1.6.2 to 1.4 give a bad audio to calling party (words are cutted, you can't understand the words). On callee party it's still good. We replace 1.6.2 version with 1.4.38 and everything is going back to normal, good audio on both side does'nt matter who call. I already opened another thread about problem with iax and Asterisk 1.6.2 (rsa auth not working anymore). Are there some known problems with iax and 1.6 version of Asterisk? Thanks for any hint -- Daniel -- _ -- 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 1.6 iax auth rsa failed with policie not found
Hi, I had 2 Asterisk servers connected together in iax with auth=rsa and proper keys for user and peer in each direction. It worked well till I upgraded one of them to Asterisk 1.6.13 Since I get No authority found I thought that problem came from keys as the server with 1.6.13 was changed in the mean time, so I regenerated both keys on each server and copy the public of each one to the other: problem stays. What am I missing? What changes in 1.6 where made concerning this matter? Thanks for any hint. -- Daniel -- _ -- 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] Call sip:u...@domain.com?
Le 17/12/2010 07:45, Gilles a écrit :
On Thu, 16 Dec 2010 17:05:35 -0500, Jamie A. Stapleton
jstaple...@computer-business.com wrote:
Just add something like this to your dialplan:
exten=1234,1,Dial(SIP/u...@domain.com)
Then, when you dial 1234 on your XLite, it will connect you to u...@domain.com.
Thanks Jamie, but isn't there a universal way to solve this, so that
users can dial any SIP number without first having to create an
extension for that specific number?
Then create a prefix for SIP calls
exten=_9.,1,Dial(SIP/${EXTEN:1})
and you dial 9u...@domain.com from XLite
Remember that calling sip URL is not as easy with a phone. Imagine you have an
ATA with DECT or POTS
phone connected on it: how to send alpha characters or @ ?
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Re: [asterisk-users] Call sip:u...@domain.com?
Le 17/12/2010 12:48, Gilles a écrit :
On Fri, 17 Dec 2010 10:16:00 +0100, Administrator TOOTAI
ad...@tootai.net wrote:
Then create a prefix for SIP calls
exten=_9.,1,Dial(SIP/${EXTEN:1})
and you dial 9u...@domain.com from XLite
Remember that calling sip URL is not as easy with a phone. Imagine you have an
ATA with DECT or POTS
phone connected on it: how to send alpha characters or @ ?
Thanks Daniel. I added that line above, told Asterisk to reload the
dialplan, and typed the following in XLite:
9*031...@ekiga.net
This is to perform an echo test
http://wiki.ekiga.org/index.php/Fun_Numbers
I guess something else must be done to Asterisk for this to work:
==
CLI
-- Executing [9*031...@my-phones:1] Dial(SIP/6011-00a1b67c,
SIP/*031600) in new stack
[Dec 17 11:43:14] WARNING[306]: chan_sip.c:2923 create_addr: No such
host: *031600
[...]
Domain part disappear.
exten=_9*.,1,Dial(SIP/${EXTEN:1...@ekiga.net)
In Xlite call 9*031600
You should read info on voip.org to learn basis of Asterisk.
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Re: [asterisk-users] Call sip:u...@domain.com?
Le 17/12/2010 16:52, Gilles a écrit :
On Fri, 17 Dec 2010 14:36:58 +0100, Administrator TOOTAI
ad...@tootai.net wrote:
Domain part disappear.
exten=_9*.,1,Dial(SIP/${EXTEN:1...@ekiga.net)
In Xlite call 9*031600
Thanks for the tip but I wanted to be able to call _any_ SIP number,
not just Ekiga, so needed a destination-agnostic solution.
You can use SipBroker.
http://www.sipbroker.com/sipbroker/action/providerWhitePages
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Daniel
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Re: [asterisk-users] Asterisk + VOSP account working configuration?
Le 15/12/2010 15:21, Gilles a écrit : [...] ;IMPORTANT: outgoing must be BEFORE incoming [vosp_outgoing] type=peer host=myvosp.com username=myaccount secret=mypasswd fromuser=myaccount fromdomain=myvosp.com nat=yes canreinvite=no [vosp_incoming] type=peer host=myvosp.com nat=yes canreinvite=no context=from_vosp Why 2 context? Todays Asterisk versions only needs one peer context for incoming/outgoing. Something like [vosp] type=peer host=myvosp.com username=myaccount secret=mypasswd fromuser=myaccount fromdomain=myvosp.com nat=yes canreinvite=no context=from_vosp -- Daniel -- _ -- 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] Atcom IP-4B ISDN IP PBX?
Le 13/12/2010 11:43, Gilles a écrit : [...] In case someone from France follows this thread, I'm interested in any feedback about professional-grade ADSL that supports VoIP, as a serious alternative to ISDN for telephony We are selling our own xDSL but a France Telecom Pro can do the job. Always dedicate the ADSL line to VoIP, use the right codec and you will have the quality you need. In big towns, some of our cutomers uses ADSL from Free Telecom without any problem. Anyway, we are today more and more upgrading our customers Internet access from ADSL to SDSL which become affordable and let us the possibility to use video during calls. -- Daniel -- _ -- 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] HA8 cards and RED alarm
Le 05/12/2010 20:28, Olivier a écrit : [...] Which Dahdi version ? I had to use latest trunk to have mine working. Thanks for your reply SrvPhone2*CLI dahdi show version DAHDI Version: 2.4.0 Echo Canceller: FYI I got it: cable was defect. -- Daniel -- _ -- 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] HA8 cards and RED alarm
Hi,
I have 2 servers: one is running 2 B410P cards with 8 euroisdn lines
(mISDN) connected on it, everything runs fine.
I prepare a new server - HP 360 G8- with 2 HA8 cards each of them 1
module of 4 lines. Already had with this machine an RMA on both cards as
they was faulty and crashed the server.
What happends is that when I connect cables on the HA8 modules (those
cables are unpluged from working server and connected to the new one)
nothing happend on the dahdi status, alarm is RED. Two days ago one
cable changed his staus to YELLOW (?) and then became again RED.
Below are relevant outputs. I created those config files with one of the
previous card which worked a short time and it was OK.
Could it be possible that modules have also to go for RMA?
Thanks for any hint.
SrvPhone2:/etc/asterisk# cat chan_dahdi.conf
;
; DAHDI telephony interface
;
; Configuration file
;
; You need to restart Asterisk to re-configure the DAHDI channels
; CLI reload chan_dahdi.so
; will reload the configuration file,
; but not all configuration options are
; re-configured during a reload.
[channels]
;
; Default language
;
language=fr
;
; Default context
;
context=isdn
internationalprefix = 00
nationalprefix = 0
usecallerid=yes
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
switchtype=euroisdn
;
; Allow call parking
; ('canpark=no' is overridden by 'transfer=yes')
;
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
;
; If you are having trouble with DTMF detection, you can relax the DTMF
; detection parameters. Relaxing them may make the DTMF detector more
likely
; to have talkoff where DTMF is detected when it shouldn't be.
;
;relaxdtmf=yes
;
; You may also set the default receive and transmit gains (in dB)
;
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
immediate=no
; This feature can also easily detect false hangups. The symptoms of this is
; being disconnected in the middle of a call for no reason.
;
callprogress=yes
progzone=be
; For fax detection, uncomment one of the following lines. The default
is *OFF*
; We use NVFaxDetect stuff for this
;
;faxdetect=both
;faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no
group=1
signalling=bri_cpe
context=isdn
channel = 1,2,4,5,7,8,10,11,13,14,16,17,19,20,22,23
SrvPhone2:/etc/dahdi# cat system.conf
loadzone = be
defaultzone = be
span = 1,1,0,ccs,ami
bchan = 1,2
hardhdlc = 3
span = 2,2,0,ccs,ami
bchan = 4,5
hardhdlc = 6
span = 3,3,0,ccs,ami
bchan = 7,8
hardhdlc = 9
span = 4,4,0,ccs,ami
bchan = 10,11
hardhdlc = 12
span = 5,5,0,ccs,ami
bchan = 13,14
hardhdlc=15
span = 6,6,0,ccs,ami
bchan = 16,17
hardhdlc = 18
span = 7,7,0,ccs,ami
bchan = 19,20
hardhdlc = 21
span = 8,8,0,ccs,ami
bchan = 22,23
hardhdlc = 24
SrvPhone2*CLI dahdi show status
Description Alarms IRQbpviol CRC4
Fra Codi Options LBO
HB8- Board 1 RED 0 0 0
CCS AMI 0 db (CSU)/0-133 feet (DSX-1)
HB8- Board 1 RED 0 0 0
CCS AMI 0 db (CSU)/0-133 feet (DSX-1)
HB8- Board 1 RED 0 0 0
CCS AMI 0 db (CSU)/0-133 feet (DSX-1)
HB8- Board 1 RED 0 0 0
CCS AMI 0 db (CSU)/0-133 feet (DSX-1)
HB8- Board 2 RED 0 0 0
CCS AMI 0 db (CSU)/0-133 feet (DSX-1)
HB8- Board 2 RED 0 0 0
CCS AMI 0 db (CSU)/0-133 feet (DSX-1)
HB8- Board 2 RED 0 0 0
CCS AMI 0 db (CSU)/0-133 feet (DSX-1)
HB8- Board 2 RED 0 0 0
CCS AMI 0 db (CSU)/0-133 feet (DSX-1)
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Daniel
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Re: [asterisk-users] 2 HB8 cards in one server - first one is not recognized, the second is
Le 26/10/2010 14:49, Shaun Ruffell a écrit : [...] First, Digium technical support would be more than happy I'm sure to help you trouble shoot this. That being said... First thing I would do is update to the current trunk of dahdi-linux. Revision 9397 [1] http://svn.asterisk.org/view/dahdi?view=revisionrevision=9397 was added because of some systems that did not provide reliable polling from the board side, which could result in erroneous your firmware may be corrupted... messages. However, since you have one card that works and one that doesn't I give this a low probability of fixing it. Installed trunk from today. Same result. Next, if updating the driver does not help and if the problem follows the card (i.e., you can swap cards and now the second card fails to load), I would disable dahdi from starting automatically, power off your system, remove the working card, power on, and try modprobe wctdm24xxp forceload=1 on the chance that the firmware on the board actually is corrupted. [ 689.968684] wctdm24xxp :0b:08.0: PCI INT A - GSI 31 (level, low) - IRQ 31 [ 692.011458] wctdm24xxp :0b:08.0: Timeout waiting for receive frame. [ 692.015668] wctdm24xxp :0b:08.0: firmware: requesting dahdi-fw-hx8.bin [ 692.039148] wctdm24xxp :0b:08.0: Reloading firmware. Do not power down the system until the process is complete. [ 694.055980] wctdm24xxp :0b:08.0: Timeout waiting for receive frame. [ 694.056052] wctdm24xxp :0b:08.0: Hx8 firmware version: 1.128 [ 694.078435] wctdm24xxp :0b:08.0: PCI INT A disabled [ 694.078443] wctdm24xxp: probe of :0b:08.0 failed with error -5 If neither of those things work, you may need to RMA your card. I will do it. BTW, does someone use Digium cards -specially HB8- in HP 360 G6 servers? I have some doubts about compatibility of those machine and telephony cards. Regards -- Daniel -- _ -- 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] 2 HB8 cards in one server - first one is not recognized, the second is
Le 26/10/2010 14:49, Shaun Ruffell a écrit : On 10/26/2010 06:38 AM, Administrator TOOTAI wrote: I installed 2 HB8 cards each of them with a Quad Bri modules in a HP 360 G6 running Debian Squeeze. Here is an output of dmesg wafter server has booted: [...] before asking RMA for the card, I would like to know what you think about this matter. First, Digium technical support would be more than happy I'm sure to help you trouble shoot this. That being said... First thing I would do is update to the current trunk of dahdi-linux. Revision 9397 [1] http://svn.asterisk.org/view/dahdi?view=revisionrevision=9397 was added because of some systems that did not provide reliable polling from the board side, which could result in erroneous your firmware may be corrupted... messages. However, since you have one card that works and one that doesn't I give this a low probability of fixing it. Didn't test this yet but Next, if updating the driver does not help and if the problem follows the card (i.e., you can swap cards and now the second card fails to load), switching cards gives kernel panic :-( on boot I would disable dahdi from starting automatically, power off your system, remove the working card, power on, and try modprobe wctdm24xxp forceload=1 on the chance that the firmware on the board actually is corrupted. Will try card by card, then slot per slot Thanks for your help -- Daniel -- _ -- 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] Checking SIP Headers existence and content
Le 05/10/2010 05:13, VoIP Question a écrit :
Hello,
Hi
I would like to verify if a specific SIP header exists, and if yes,
extract the partial content from another header.
1. Is there a way to verify if a specific header exists?
2. How do I extract data that is between the first : and the following
@? Specifically, The data looks like sip:1234567...@10.0.0.1:5060
http://sip:1234567...@10.0.0.1:5060 and I would like to get only
the 1234567890
Something like
exten = s,1,Set(__DIALEDNUMBER=${SIP_HEADER(TO):5})
exten = s,n,Set(__DIALEDNUMBER=${CUT(DIALEDNUMBER,@,1)})
exten = s,n,GotoIf($[${DIALEDNUMBER:0:1} != +]?numberIsOK)
exten = s,n,Set(__DIALEDNUMBER=${CUT(DIALEDNUMBER,+,2)})
Take a look here
http://www.voip-info.org/wiki/view/Asterisk+func+sip_header
voip-info.org should be in your favorites ;-)
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Daniel
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Re: [asterisk-users] SMS and fixed land lines
Le 06/09/2010 15:10, Olivier a écrit : Hi, Hello 1. Do you have any experience with receiving incoming SMS on an analog or ISDN landline ? How can then you differentiate an SMS call from a voice call ? From http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms it seems the way to tell an inbound call is an SMS one is to read the callerid number but does this still apply with calls coming from cellphones ? 2. Is SMS service compatible with PRI lines ? As stated by Philipp, SMSC is unique. However -in France at least- SMS sended to landlines are altered and sended as voice messages by the operators. For messages from Orange you will recognize that's a SMS as the callerID is the Orange SMSCs one. For SFR no luck, Bouygues don't tested. -- Daniel -- _ -- 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] SMS and fixed land lines
Le 06/09/2010 17:39, Olivier a écrit : 2010/9/6 Administrator TOOTAI ad...@tootai.net mailto:ad...@tootai.net Le 06/09/2010 15:10, Olivier a écrit : Hi, Hello 1. Do you have any experience with receiving incoming SMS on an analog or ISDN landline ? How can then you differentiate an SMS call from a voice call ? From http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms it seems the way to tell an inbound call is an SMS one is to read the callerid number but does this still apply with calls coming from cellphones ? 2. Is SMS service compatible with PRI lines ? For SFR no luck, What do you mean by that ? That SMS from cellphones cannot reach landlines or are not using a unique SMSC callerid which makes them unrecognizable ? No unique SMSC. In the voice message they send you, it's You receive an SMS from John Doe, press 1 if you want to listen the message Very funny when you have your voicemail activated or fax detection before voice :-( The callerID is the one from the SMS sender but this means nothing as you can send SMSs from a ... landline! They are so stupid ... -- Daniel -- _ -- 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] SMS and fixed land lines
Le 06/09/2010 19:31, Randy R a écrit : [...] Some of this may have changed, but when I has asterks and a fixed-line SMS service from France Télécom, that's the way it worked. End of 2009 SMS sended to landlines where easy to treat, we even setup an SMS2Mail gw. Those days, we only treat SMSs from Orange/France Telecom as they SMSC has is own callerID. -- Daniel -- _ -- 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] sending sms from Asterisk server
Le 18/08/2010 16:03, Tino a écrit : Hello Johann, Thanks for your advice in this matter. But i am not sure how to pass the numbers to be sent sms in the dialplan. agi(script,param1,param2,...,paramX) from your dialplan where script lies in /var/lib/asterisk/agi-bin On Wed, Aug 18, 2010 at 3:13 AM, Johann Hoehn johann.ho...@ecommerce.com mailto:johann.ho...@ecommerce.com wrote: On 08/17/2010 09:00 AM, Tino wrote: Hello, I would like to send sms to some external phone numbers from my asterisk server. Is it possible to send sms via softphones like X-Lite ? . Any tips regarding this will be helpful thanks This is easy to do by using email to SMS gateways. A list of them is on wikipedia (http://en.wikipedia.org/wiki/List_of_SMS_gateways). For the Asterisk side, you have an extension that sends the email. I personally use an AGI script for this part, but you could use a System() call as well. -- Daniel -- _ -- 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] Peculiar Polycom IP6000 behavior
Hello Le 27/07/2010 20:57, Cassius Smith a écrit : Here's a strange thing. I'm deploying Asterisk 1.6.2.9 with a pile of Cisco 79xx phones. For conference rooms we're using Polycom IP6000's. We bought two of them brand new. [...] Any ideas? I'm stumped. If tour register server is outside your local network, you will have a problem as the IP [5|6|7]000 are registering using port 5060 on public IP (symetric nat) which will allow only one device. -- Daniel -- _ -- 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] Register Attacks End of ENUM ?
Le 25/07/2010 02:11, Norbert Zawodsky a écrit : Hello again! Hi after it being relatively quiet her for the last weeks, my Astrerisk server was the target of 3 of that nasty REGISTER attacks during the last days. [...] Do like most of us are acting: use fail2ban. -- Daniel -- _ -- 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] How to deal with voice SMS - Asterisk 1.4
Hi list, I face a problem with voice SMSs. In some countries, if you send an SMS to a landline number, the mobile operator will record the message and then call this number. When picking up the phone you hear You get an SMS from phone number, press 1 to listen the message, 2 to repeat the sender phone number. If you press 1 you hear the message and after it you have the possibility to press 1 to repeat message or 2 to repeat the sender phone number. In a perfect world it's OK, but not here. There is fax detection, users send to voicemail directly, companies message with open hours aso. How to treat this with Asterisk knowing that such kind of messages are sended -at least for some operators- with a special callerID? At this time, I check the length of the first message, record it message for length seconds, then sendDTMF(1) and go to voicemail. All is good except that: - if first message length is changed it's no more working - same if the behaviour is changed (eg press 2 or 3 or ...) - but more of all, the callee never know the phone number from the person who send the message Is it possible to know the voicemail file name just being recorded? In this case, I could merge my first recorded file with the one of the voicemail. Other solution? How do you guys are handling such situation? -- Daniel -- _ -- 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] How to deal with voice SMS - Asterisk 1.4
Le 15/07/2010 10:38, Gordon Henderson a écrit : On Thu, 15 Jul 2010, Administrator TOOTAI wrote: Hi list, I face a problem with voice SMSs. In some countries, if you send an SMS to a landline number, the mobile operator will record the message and then call this number. When picking up the phone you hear You get an SMS fromphone number, press 1 to listen the message, 2 to repeat the sender phone number. If you press 1 you hear the message and after it you have the possibility to press 1 to repeat message or 2 to repeat the sender phone number. In a perfect world it's OK, but not here. There is fax detection, users send to voicemail directly, companies message with open hours aso. How to treat this with Asterisk knowing that such kind of messages are sended -at least for some operators- with a special callerID? BT in the UK use a specific caller ID when speaking SMS messages to you - however you still have 2 choices - you can listen to the message as spoken by BT's Digital Dot, or if you instrict them, they'll send the message digitally (over the analogue line as FSK tones) so that compatable equipment can then display the original message text (e.g. Siemens DECT phones) Which means I have to ask them for each landline I'm taking care ... No chance. [...] So I doubt there will be a universal solution. I agree even if it's not what is was expecting ;-) If your country supports switching to FSK sending, then it might be worth while investigating the SMS application and doing it all digitally. You could then setup a local number to email map and email the message rather than try to speak it. I got it work with smsq and it worked well BT's system does produce some intereting results... Eks eks eks Ell Oh Ell. ;-) :-) -- Daniel -- _ -- 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] one for your filters
Le 23/06/2010 21:28, Gordon Henderson a écrit : [...] I'd like to have a look, but can't - I think there may be issues with your registrar for your domain - from where I am, there are no glue records for the nameservers, therefore I can't look it up... Looks like it was last edited just over 4 weeks ago, so maybe some caches are starting to time-out... From whois: Domain servers in listed order: DOMAIN0.SEDWARDS.COM DOMAIN1.SEDWARDS.COM You need to supply the IP address of the nameservers (the glue records) if they're inside your own domain... (sorry to post this to the list, but I can't email you because of this - looks like you're still getting list traffic though!) Same here, also from Europe. -- Daniel -- _ -- 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] Connecting 1-2 GSM ports to asterisk?
Le 21/05/2010 16:19, Motiejus Jakštys a écrit : Hi, List, I am looking for a cheapest (and therefore most funny) way to attach GSM card to my asterisk home box. Have a look at chan_mobile (bluetooth connection) -- Daniel -- _ -- 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] Being attacked by an Amazon EC2 ...
Gordon Henderson a écrit : Just a heads-up ... my home asterisk server is being flooded by someone from IP 184.73.17.150 which is an Amazon EC2 instance by the looks of it - they're trying to send SIP subscribes to one account - and they're flooding the requests in - it's averaging some 600Kbits/sec of incoming UDP data or about 200 a second )-: This is much worse than anything else I've seen. List of Amazon IP's from which we already have been attacked on several of our servers in Europe (blocked with Fail2Ban): 75.101.195.70 79.125.30.56 184.72.6.92 184.73.70.8 184.73.21.31 184.73.16.184 204.236.169.224 We also faced attack from China, Germany, Romania, Israel and Palestine -- Daniel -- _ -- 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] State of 64 bits applications in Asterisk
Hi, what is the state at this time for 64bits applications and compatibility with 1.6.2 Mainly speaking about FFA, SFA, G729. Thanks for any information -- Daniel -- _ -- 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] Server response time
Juan C. Villa a écrit : [...] The total lag from Germany to USA (2 way) is around ~110ms (Just tested it today). Who this cause any issues with my VoIP applications? Right now I have two VoIP boxes installed in Switzerland which are connected to my server in California (avg response time = 190ms) and I have no problems at all. What would you guys advice? FYI, I made an mtr to the IP 143.215.103.174, one from one of our servers in Switzerland, the second from an Hetzner one: both give 112 ms AVG time. Regards -- Daniel -- _ -- 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] Server response time
Juan C. Villa a écrit : Hey Guys, HI Juan I am considering leasing a new server in Germany to run my Asterisk infrastructure and I was wondering how response time would affect the performance of the system. Right now I have a response time of around 60-70ms with my server in California. The server in Germany would have a response time of around 140ms (both ways). My DID/Termination providers are in Canada and the USA, and all my voip boxes are also in the USA. Any suggestions or recommendations? I'm in Europe and had used Boadvoice few years ago. I stopped because of the bad quality due to latency. Last year I bought a 20 Skype seat at Gizmo but never could use them: latency Europe - US - Europe + Skype network was a total non sense and never could have a acceptable voice quality. You could do it if people to connect where US/Europe and vice versa. -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registering of Asterisk against a SIP provider
Hi Daniel Bareiro a écrit : [...] Hours ago the IP changed and the domain was updated satisfactorily, but in spite of this I was obtaining the registering failures that I mentioned above. After to restart Asterisk (1.4.24.1), I no longer had this problem of registering. But there would be some way to solve this problem? [...] It's an old story. Asterisk check DNS when it start that's why it's ok after you have it restarted. When I was running Asterisk using dynamic addresses, I made following: - modify sip.conf to include a file placed where ever you want, contents being externalip/externalhosts and all others info needed related to external IP - restarted myself ADSL line with a cron script each night - this script extract/found the new IP using the method you prefer (eg ping your dyndns host until response and than you have your new IP and insert the IP in the file you include in sip.conf - this script restart asterisk and voila :-) Was working like a charm. -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.2 : global vars not read/set after #include w/ globals
sean darcy a écrit : [...] Context names cannot be duplicated, unless you suffix them with (+) to allow them to be added together. It does not matter whether it is the 'global' context or any other context. Well Dialplan reloaded. == Parsing '/etc/asterisk/extensions.conf': == Found .. == Parsing '/etc/asterisk/exts/gvoice.exten.conf': == Found cat exts/gvoice.exten.conf [+globals] test-global = need-a-plus-sign . but no test-global in dialplan show globals :( should be [globals](+) -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unregistred users can pass calls, peer being static
Hi, we had an attack on a server and we don't understand how it was possible, Asterisk 1.4.28/Debian Lenny 5.1 Attacker came from PALTEL, network 188.161.128.0/18 Hacked account had following setup: [111] type=friend username=111 context=from-111 host=11.22.33.44 dtmfmode=auto qualify=yes nat=yes canreinvite=no defaultip=11.22.33.44 port=35060 disallow=all allow=ulaw,alaw call-limit=2 Despite this, I saw in my logs that someone hacked this account and could place calls! in logs we have: [Jan 27 04:00:13] ERROR[29715] chan_sip.c: Peer '111' is trying to register, but not configured as host=dynamic [Jan 27 04:00:13] NOTICE[29715] chan_sip.c: Registration from 'sip:1...@ourasteriskip' failed for '188.161.152.245' - Peer is not supposed to register [Jan 27 04:00:18] VERBOSE[30669] logger.c: -- Executing [972599400...@from-111:1] NoOp(SIP/111-16eb, Incoming call from ) in new stack As you see 111 could place a call even having not registered, which he is not supposed to do. How is this possible? -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unregistred users can pass calls, peer being static
wins mallow a écrit : On Wed, 2010-01-27 at 11:47 +0100, Administrator TOOTAI wrote: [...] Check your sip.conf allowguest=no Guest are allowed and going to a different context. Logs are showing that calls are going out to the from-111 context, so its this account which was hacked. Thanks for your answer. -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unregistred users can pass calls, peer being static
Olle E. Johansson a écrit : 27 jan 2010 kl. 11.47 skrev Administrator TOOTAI: Hi, we had an attack on a server and we don't understand how it was possible, Asterisk 1.4.28/Debian Lenny 5.1 Attacker came from PALTEL, network 188.161.128.0/18 Hacked account had following setup: [111] type=friend username=111 context=from-111 host=11.22.33.44 dtmfmode=auto qualify=yes nat=yes canreinvite=no defaultip=11.22.33.44 port=35060 disallow=all allow=ulaw,alaw call-limit=2 Despite this, I saw in my logs that someone hacked this account and could place calls! in logs we have: [Jan 27 04:00:13] ERROR[29715] chan_sip.c: Peer '111' is trying to register, but not configured as host=dynamic [Jan 27 04:00:13] NOTICE[29715] chan_sip.c: Registration from 'sip:1...@ourasteriskip' failed for '188.161.152.245' - Peer is not supposed to register [Jan 27 04:00:18] VERBOSE[30669] logger.c: -- Executing [972599400...@from-111:1] NoOp(SIP/111-16eb, Incoming call from ) in new stack As you see 111 could place a call even having not registered, which he is not supposed to do. How is this possible? [...] type=friend creates two objects in your asterisk server, one peer and one user. Asterisk primarily match the user objects for incoming calls on the From: username. In this case, you have 111 as the username (regardless of the username field which is not the username btw). You have no secret defined, so anyone placing a call from a URI that has 111 as the username part will be able to use your server. Calling from sip:1...@asterisk.org as well as sip:1...@mydomain.com will work without authentication - from any IP address out there. Very poor security indeed. 1) Add a secret. 2) Add ACL rules (permit/deny) to restrict IP address access 3) Change to type=peer and we'll only match on IP for incoming calls. I still recommend using authentication. So the fact that host is setted to an IP doesn't matter in case of type=friend. Didn't notice that, thanks for the explanation. [..] Make sure you read this and act upon it! Sure, already done. Thanks for your answer. -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unregistred users can pass calls, peer being static
Hi Kevin Kevin P. Fleming a écrit : [...] This conversation brings to mind two possible ways we could improve Asterisk to help users from falling into this trap: 1) When a sip.conf entry is defined as 'type=friend' *and* has a specific host IP address (not dynamic), we could just ignore the 'user' part and create only the 'peer' part. This would result in incoming calls being matched by IP address instead of username, which is likely what the administrator wants anyway. 2) Alternatively, if people really do want both the 'user' and 'peer' objects to exist, then we could automatically put an ACL on the 'user' object that restricts access to it to only the defined IP address. This also could apply to dynamic hosts, but only those that are defined without a secret (no authentication required), which seems like a terrible configuration and we don't really need to do anything to make it work 'better' :-) #1 sounds great for me. Don't know for others but for us SIP EP are mainly setted as user host=dynamic+secret or host=IP address meaning permit only this IP. Other solution would be -in case of host=IP address- to set permit=IP address/32 deny=0.0.0.0/0.0.0.0 if those parameters are *not* present All of those solution are compatible with the fact that information should be given if the case appear. -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: Inbound South America numbers
Hi, is someone able to provide inbound DID for South America, at least Bolivia, Colombia, Panama and Venezuela. Please contact me of list, thanks Regards -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitoring SIP Skype connections
Myles Wakeham a écrit : [...] Are there tools or add-ons available for this that will email me when a SIP registration goes offline? Any suggestions for this would be greatly appreciated. Hi Myles, first, best wishes to the list for this new 2010 year. To answer your question, you can run a cron job each x minutes -supposing that you qualify your provider- launching a script like #!/bin/bash isOffLine=`/usr/sbin/asterisk -rx 'sip show peers'| grep MySIPProvider | grep OK` if [ $isOffLine = ]; then # start what you want, for instance do a sip reload fi exit 0 -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AsteriskNow and language
Administrator TOOTAI a écrit : Hi, I installed AsteriskNow and upgraded FreePBX to 2.6.0. In a sip extension definition, when I set language, it is not reported in the extensions_custom.conf file (eg language=xx). Am I missing something or is it not the right way to set language? Hello, sorry to insist on this, does nobody use AsteriskNow? I register to the AsteriskNow mailing list, no more luck to get answer. I also notice that call-limit was setted to 50! Where can I modify thos options. Thanks for any hint. Merry christmas -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AsteriskNow and language
Hi, I installed AsteriskNow and upgraded FreePBX to 2.6.0. In a sip extension definition, when I set language, it is not reported in the extensions_custom.conf file (eg language=xx). Am I missing something or is it not the right way to set language? BTW, is this a valid place for AsteriskNow questions? Dedicated mailing list seems dead. Thanks for answer -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] NvFaxdetect and Asterisk 1.4.27 - Someone get it work?
Hello, I had an 1.4.21-2 Asterisk running on Debian/Etch with app_nv_faxdetect running on it without any problem. I upgraded the server to Debian/Lenny and Asterisk 1.4.27 and app_nv_fax_detect is not working anymore: on an incoming call, application is launched and never exit :-( I reinstalled 1.4.21-2 compiled against the new environment and get the same result! Lenny is 64bits version, 2.6.26-2-amd64 stock Does someone get NVFaxDetect work with latest Asterisk 1.4 version? Thanks for any hint -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
Lee Howard a écrit : In your sip.conf file allowguest defaults to yes. This means that anyone that can reach the SIP ports on that system has access to make unauthenticated calls, by default. The administrator actually has to go in and turn it off to prevent unauthenticated SIP calls (in whatever context [general] points at). Does anyone else agree with me that this is a poor default? I'd like to see the default setting changed. It seems to me that this default is the reason behind the doc/security.txt bias against using the default context for toll calls. Agree. Another possibility would be to have a guestcontext defined in default. This context would exist but empty. -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphone in Web
ABBAS SHAKEEL a écrit : Hello Hi I am thinking to develop a softphone that is integrated into web.(in form of APPLET or some thing else) Ie a user with with just a PC with Net Browser(fire fox etc) Installed can make call.. Is there some thing developed before like this that is open source ?? Take a look at Mozphone -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SFA - No channel cause 66
Hi, after having tested SFA in august, I didn't use it for some times and now I receive the subject error when calling through Skype channel. Has anyone an idea on what can be the problem? Thanks -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call-limit on dahdi channel
Alex Samad a écrit :
Hi
how do i set the call-limit on a dahi line - its connected to the pstn
network - shared fax line. How do i tell asterisk not to send more than
1 call there !
exten = _XXX.,20(Start),Set(GROUP()=PSTN)
exten = _XXX.,n,GotoIf($[${GROUP_COUNT(PSTN)}=0]?lineOpen)
exten = _XXX.,n,Congestion()
exten = _XXX.,n,Hangup(34)
exten = _XXX.,n(lineOpen),NoOp(Place your call to DAHDI channel)
--
Daniel
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[asterisk-users] invalid extension
Hello, with Asterisk 1.6.1.6 I try to hangup a call if called extension is not existing. For this purpose I would use the internal i extension but seems not to work. [MyContext] exten = s,1,NoOp(Call is treated as it should) exten = s,n,NoOp(next step) exten = s,n,NoOp(aso ...) exten = _[a-zA-Z].,1,Goto(s,1) ; accept exten LEN 1 alpha exten = _X.,1,Goto(s,1); accept exten LEN 1 numeric exten = i,1,NoOP(sorry, extension doesnt exist) ; all 1 digits exten exten = i,n,Hangup ; refused, end of call What I have when calling a one digit extension -in this case h- is: == Using SIP RTP CoS mark 5 [Sep 7 18:51:03] NOTICE[6084]: chan_sip.c:18523 handle_request_invite: Call from '' to extension 'h' rejected because extension not found. == Using SIP RTP CoS mark 5 Should it not go to i extension? If I call the i or s extension it's going well. Am I missing something? -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] invalid extension
Miguel Molina a écrit : [...] The 'i' extension only works in applications like Background(), WaitExten() and everything that uses DTMF to route extensions within a context. Well, from reading voip.org it's not really clear than ... [...] Because the call is not accepted there's no need for a hangup (in a SIP environment). Well, I like when logs are clear ;-) and not have to guess :-) If you want to explicitly hangup calls using the dialplan, for your case add a one-digit catch all and leave your good calls with a 2-digit minimum: exten = _X,1,NoOP(sorry, extension doesnt exist) ; all 1 digits exten exten = _X,n,Hangup Did it but get 2 hangup! First calling 2...@domain.local == Using SIP RTP CoS mark 5 -- Executing [...@from-guest:1] Goto(SIP/sip.tootai.net-084b1dc8, h,1) in new stack -- Goto (from-guest,h,1) -- Executing [...@from-guest:1] Hangup(SIP/sip.tootai.net-084b1dc8, ) in new stack == Spawn extension (from-guest, h, 1) exited non-zero on 'SIP/sip.tootai.net-084b1dc8' -- Executing [...@from-guest:1] Hangup(SIP/sip.tootai.net-084b1dc8, ) in new stack == Spawn extension (from-guest, h, 1) exited non-zero on 'SIP/sip.tootai.net-084b1dc8' Second calling h...@domain.local == Using SIP RTP CoS mark 5 -- Executing [...@from-guest:1] Hangup(SIP/sip.tootai.net-084c97b8, ) in new stack == Spawn extension (from-guest, h, 1) exited non-zero on 'SIP/sip.tootai.net-084c97b8' -- Executing [...@from-guest:1] Hangup(SIP/sip.tootai.net-084c97b8, ) in new stack == Spawn extension (from-guest, h, 1) exited non-zero on 'SIP/sip.tootai.net-084c97b8' exten = _XX.,1,Goto(s,1) ; accept exten LEN 1 numeric Here your calling a three or more digits ;-) That will be enough to hangup what you want to, adjusting it to your needs. I will leave with this :-) Many thanks for the informations. -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.4.26-2, DAHDI-2.2.0, B410P and BRI
Hello everybody, I try to install -Ubuntu 8.04 server- a B410P and a TDM2400P together with Asterisk 1.4.26-2, dahdi-linux-2.2.0.2 and dahdi-tools-2.2.0. Problem I face is the following one: CLI module load chan_dahdi.so == Registered application 'DAHDISendKeypadFacility' == Registered application 'ZapSendKeypadFacility' == Parsing '/etc/asterisk/chan_dahdi.conf': Found [Sep 4 19:07:37] ERROR[18464]: chan_dahdi.c:11675 process_dahdi: Unknown signalling method 'bri_cpe_ptmp' [Sep 4 19:07:37] ERROR[18464]: chan_dahdi.c:7677 mkintf: Signalling requested on channel 1 is FXO Loopstart but line is in ISDN PRI signalling [Sep 4 19:07:37] ERROR[18464]: chan_dahdi.c:11294 build_channels: Unable to register channel '1-2' Why signalling bri_cpe_ptmp is not recognized (for tests, pri_cpe is loading well, bri_cpe or bri_net gaves also errors)? In chan_dahdi.conf I have [...] switchtype = euroisdn signalling = bri_cpe_ptmp ;signalling = pri_cpe channel = 1-2 dahdi_scan give me for port: [1] active=yes alarms=OK description=B4XXP (PCI) Card 0 Span 1 name=B4/0/1 manufacturer=Digium devicetype=Wildcard B410P location=PCI Bus 00 Slot 12 basechan=1 totchans=3 irq=5 type=digital-TE syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI,HDB3 framing_opts=ESF,D4,CCS,CRC4 coding=AMI framing=CCS Asterisk and DAHDI are compiled with libpri as: # strings /usr/lib/asterisk/modules/chan_dahdi.so|grep 'DAHDI Tele' DAHDI Telephony Driver w/PRI DAHDI Telephony w/PRI I took a look in chan_dahdi.c and found nowhere info concerning bri. Is the bri stuff from DAHDI only working with Asterisk 1.6 branches? Should I switch to mISDN? Thanks for any hint or comments. -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gizmo Dial Out No CALLED PARTY AUDIO??
Rob a écrit : Yes ... as a matter of fact here is the sip.conf ... obviously private info removed [...] Did you try to call Gizmo numbers to see if you have success with them? ** Hear your Gizmo5 number repeated back to you. *0 Test your router's SIP compatibility. 411 The voice-activated Tellme information service. 1-747-474-ECHO 1-747-474-3246 Echo Test - Repeats back whatever you say. -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone had any luck with SIP clients on the iPhone platform?
randulo a écrit : Hi, Hello I've tried two SIP clients so far and both have unusable outgoing audio quality. [...] Anyone have any recommendations? I made few test with various client, Sip and IAX, on iPhone first generation: . frings: good quality but to much delay. Also I don't like the fact that it's Frings server which register to Asteris, not the client. Question of privacy . iSip: good quality but also delay . siax: same as above, even better quality than iSip, but still delay, doesn't matter Sip or IAX . Weephone: perfect, good sound no delay. All those tests where made from one location to the same Asterisk server somewhere on Internet. Also I didn't pay attention on the look or if you can connect few accounts. Anyway, no one of those clients have the quality of a Nokia SIP client. To notice: when always WifI connected, the iPhone start to be hot, not cool when you're always on the phone ;-) -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] original reformat extension
Karl Fife a écrit :
[...] there are times when I want to send the call to another context in its
original un-reformatted state. Naturally the ${EXTEN} variable has been
changed. It occurred to me to use CALLERID(DNID) as such:
exten = _1NXXNXX,n(fail),Goto(other-context,${CALLERID(DNID)},1)
Before goto
exten = _NXXNXX,1,Set(__DIALEDNUMBER=${EXTEN})
exten = _NXXNXX,n,Goto(1${EXTEN},1)
and then you always have the original unformated state in ${DIALEDNUMBER}
--
Daniel
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Re: [asterisk-users] Calling issue for non-extension numbers
Kayton Sapale a écrit : Hi all, HI alone :-) Thanks to the previous replies that helped me with this before, but I got side-tracked in the middle of trying to figure this out, so apologies for posting the same issue. I use a Nokia e71, with an asterisk server and am having an issue dialing certain numbers. When I attempt to dial a local number, like xxx-xxx-, I cannot connect. What shows in the asterisk debug is the following: Found peer '104' However, if I try to dial an extension that is configured on the asterisk server, the call goes through fine. When I use another device to connect the server (another nokia actually) and dial a local number like xxx-xxx-, I see this in the debug dialog: Found peer '103' [...] Looking for 6789940793 in DLPN_Free_Outbound (domain sip.speartek.com) list_route: hop: sip:1...@192.168.111.183 It appears that my device cannot connect to the server when dialing certain numbers. Does anyone have any idea about this? From what you show us above there is nothing wrong. You should better debug your dialplan, specially if DLPN_Free_Outbound context allow numbers like 6789940793. Regards -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gizmo Dial Out No CALLED PARTY AUDIO??
Rob a écrit : Hi all, Hi I'm using GIZMO with my asterisk (1.4.13) box ... I've had CALL IN for a while and it works fine I just added CALL OUT ... I have no problem with call setup ... the called party hears me ... but I can't hear them again if the call comes INTO the server both sides work fine. Looks like a nat issue: do you have nat=yes and canreinvite=no in your sip.conf for Gizmo5? -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling issue for non-extension numbers
Kayton Sapale a écrit : Thanks Daniel. It looks like I didn't paste everything into the email, but not sure if this will make a difference: No need to send agian the same datas, I cutted non relevant part in my answer. From your other mail I'm sure that your problem is dialplan related. Could you increase verbosity to 3 or 4, pass a call and check what you have in console. Also review your dialplan to check why calls to 80055511212# finish in time out. Or simply modify your dialplan with something like exten = _8005.,1,dial(SIP/104) which should make ring your e71 at ext 104 when dialing any number of 5 digits starting with 8005 from your extension 103. -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfer Issue with IAX Trunk
Doug Lytle a écrit : Lutgring, Sam wrote: I have an IAX trunk configured between 2 Asterisk servers. Everything is working great except if the caller presses # during the call. If they press # the local PBX comes on and says transferring and tries to transfer to a blank extension. Does anyone know how to turn this off? There is no extension defined for # in the dial plan. core show application dial: t- Allow the called party to transfer the calling party by sending the DTMF sequence defined in features.conf. T- Allow the calling party to transfer the called party by sending the DTMF sequence defined in features.conf. Or change the config in features.conf -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.25 and attended transfer
Marco Sambo a écrit : Hi all, I've a problem: I update my asterisk to version 1.4.25, and the attended transfer doesn't work. [...] Marco, attented transfer are broken in 1.4.25, please upgrade to 1.4.26 (see changelog). -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
