[asterisk-users] No progress tones on transferred call
Asterisk 1.4 We are experiencing an issue on transfers where no progress tones are heard by the caller: 1. Call from 1593 (SPA525G 0026998D2) to 1595 (SPA922 000B820AF). 1595 answers 2. From 1593 initiate transfer to 1597 (SPA508G 1CDF0F4A). 1595 hears MoH. 3. 1597 starts ringing and 1593 presses transfer again. MoH stops but 1595 hears no ringing When xfer is pressed and the extension is dialled: U 203.89.001.001:5060 - 121.98.001.001:1034 INVITE sip:1CDF0F4A@192.168.1.72:5060 SIP/2.0..Via: SIP/2.0/UDP 203.89.001.001:5060;branch=z9hG4bK5286810e;rport..From: C Allerid sip:1593@203.89.001.001;tag=as72616c50..To: sip:1CDF0F4A@192.168.1.72:5060..Contact: sip:1593@203.89.001.001..Call-ID: 59ba10300b9b8cb5684eba2368c90...@203.89.001.001..cseq: 102 INVITE..User-Agent: Asterisk PBX..Max-Forwards: 70..Date: Tue, 05 Jun 2012 08:05:02 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supported: replaces..Content-Type: application/sdp..Content-Length: 262v=0..o=root 3031 3031 IN IP4 203.89.001.001..s=session..c=IN IP4 203.89.001.001..t=0 0..m=audio 13728 RTP/AVP 0 3 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:1010-16..a=ptime:20..a=sendrecv.. U 121.98.001.001:1034 - 203.89.001.001:5060 SIP/2.0 100 Trying..To: sip:1CDF0F4A@192.168.1.72:5060..From: C Allerid sip:1593@203.89.001.001;tag=as72616c50..Call-ID: 59ba10300b9b8cb5684eba2368c90...@203.89.001.001..cseq: 102 INVITE..Via: SIP/2.0/UDP 203.89.001.001:5060;branch=z9hG4bK5286810e..Server: Cisco/SPA508G-7.4.9c..Content-Length: 0 U 121.98.001.001:1034 - 203.89.001.001:5060 SIP/2.0 180 Ringing..To: sip:1CDF0F4A@192.168.1.72:5060;tag=53e23c5265d60f06i0..From: C Allerid sip:1593@203.89.001.001;tag=as72616c50..Call-ID:59ba10300b9b8cb5684eba2368 c90...@203.89.001.001..cseq: 102 INVITE..Via: SIP/2.0/UDP 203.89.001.001:5060;branch=z9hG4bK5286810e..Contact: $USER sip:1CDF0F4A@192.168.1.72:5060..Server: Cisco/SPA508G-7.4.9c..Content-Length: 0 After transfer is pressed the second time there is no further SIP messages with Asterisk CLI -- Executing [s@macro-dial:12] Dial(SIP/000B820AF-2d0a, SIP/000E08D6SIP/1CDF0F4ASIP/000E08D61|20|tTwWr) in new stack -- Called 1CDF0F4A -- SIP/1CDF0F4A-2d0b is ringing -- Stopped music on hold on SIP/0026998D2-2d08 Updated sip.conf progressinband=yes This didn't make any difference I've tried calls in different directions in case it is to do with the particular phone firmware but the direction is irrelevant. Any suggestions appreciated or if you require further information please ask. -- _ -- 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] BRI Installation
On 06/05/2012 12:39 AM, Klaverstyn, David C wrote: Hi Guys, All my installs are based on PRI ISDN. I now have a site that I need to install BRI. As I have not done a BRI install before I’m wanting to get some information from the people in the know if I need to do anything special. Typically I install libpri, dahdi Linux and tools, asterisk ...and then configure dahdi as one does for the required hardware. Is the same true for BRI with the exception of the libpri? I have this feeling that I need to install some other Linux drivers or something for BRI. I’ve purchased a Digium HB8 card and I don’t see any mention of this in /etc/dahdi/modules. I’ve looked over the documentation at https://www.digium.com/en/supportcenter/documentation/viewdocs/H8 but there doesn’t seem to be anything there that tells me how to configure dahdi or asterisk. If someone could give me some direction that would be greatly appreciated. The Hx8 User's Manual (here: http://docs.digium.com/H8/hx8_series_manual.pdf) has an entire chapter on software installation and configuration, including DAHDI, libpri and Asterisk. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] attended transfer with CEL
hello, is there someone who successfully get info about attended transfer from CEL? if yes, can you post some hints/algorithm/...? scenario A - customer B - secretary C - consultant1 D - consultant2 A - B B axfer C C axfer D i need to know time B with C (consultation) time A with C time C with D (consultation) time A with D time A with everyone (full time - from start to the end of call) -- --- Marek Cervenka === -- _ -- 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 1.8.13.0 Now Available
When will this be available at packages.asterisk.org? Thanks! EKG -- _ -- 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] OOh323 log fills with : In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
We have an Ast 1.6 installation which is connected to an Avaya using ooh323.
Something is causing the log to fill with In ooEndCall call state is -
OO_CALL_CLEAR (incoming, ooh323c_1) messages every 100ms. This causes the log
to grow to 300MB in just 5 minutes, which eventually overloads the box.
Looking through the ooh323 log below, I suspect this stems from the
Error:Failed to enqueue ReleaseComplete message to outbound queue.(incoming,
ooh323c_1) message - but we don't don't see enough H323 installations to dig
deeper. Can someone offer some suggested causes and resolutions?
Thanks!
-Date 06/01/12-
02:45:33:447 New connection at H225 receiver
02:45:33:447 Created a new call (incoming, ooh323c_1)
02:45:33:463 Receiving H.2250 message (incoming, ooh323c_1)
02:45:33:463 H.2250 message length is 12
02:45:33:463 Received Q.931 message: (incoming, ooh323c_1)
02:45:33:463 Received H.2250 Message = {
02:45:33:463 protocolDiscriminator = 8
02:45:33:463 callReference = 0
02:45:33:463 from = originator
02:45:33:463 messageType = 7d
02:45:33:464 Cause IE = {
02:45:33:464Q931NormalUnspecified
02:45:33:464 }
02:45:33:464 No UserUser IE found in ooDecodeUUIE
02:45:33:464 Error:Failed to decode received H.2250 message. (incoming,
ooh323c_1)
02:45:33:464 Decoded Q931 message (incoming, ooh323c_1)
02:45:33:464 }
02:45:33:464 ERROR:Failed ooH2250Receive - Clearing call (incoming, ooh323c_1)
02:45:33:464 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:33:464 Building Release Complete message to send(incoming, ooh323c_1)
02:45:33:464 Built Release Complete message (incoming, ooh323c_1)
02:45:33:479 Asn1Error: -4 at ooh323c/src/encode.c:584
02:45:33:479 ERROR: UserInfo encoding failed
02:45:33:479 Error:Failed to encode uuie. (incoming, ooh323c_1)
02:45:33:479 Error:Failed to encode H225 message. (incoming, ooh323c_1)
02:45:33:479 Error:Failed to enqueue ReleaseComplete message to outbound
queue.(incoming, ooh323c_1)
02:45:33:479 Receiving H.2250 message (incoming, ooh323c_1)
02:45:33:479 H.2250 message length is 12
02:45:33:479 Received Q.931 message: (incoming, ooh323c_1)
02:45:33:479 Received H.2250 Message = {
02:45:33:479 protocolDiscriminator = 8
02:45:33:479 callReference = 0
02:45:33:479 from = originator
02:45:33:479 messageType = 7d
02:45:33:479 Cause IE = {
02:45:33:479Q931NormalCallClearing
02:45:33:479 }
02:45:33:479 No UserUser IE found in ooDecodeUUIE
02:45:33:479 Error:Failed to decode received H.2250 message. (incoming,
ooh323c_1)
02:45:33:479 Decoded Q931 message (incoming, ooh323c_1)
02:45:33:479 }
02:45:33:479 ERROR:Failed ooH2250Receive - Clearing call (incoming, ooh323c_1)
02:45:33:479 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:33:579 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:33:679 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:33:779 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:33:879 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:33:979 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:34:079 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:34:179 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:34:279 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:34:379 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:34:479 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:34:579 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:34:679 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:34:779 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:34:879 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:34:979 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:35:079 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:35:179 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:35:279 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:35:379 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:35:479 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:35:579 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:35:679 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:35:779 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:35:879 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:35:979 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:36:079 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:36:179 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:36:279 In
[asterisk-users] CDRs on multiple servers.
Hello guys, I need to be able to throw cdrs on more than one servers at a time. Please let me know how this can be done. Thanks -- _ -- 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] CDRs on multiple servers.
The easiest way for you to use MySQL-Relay or MySQL-Proxy with ODBC. Regards, Faisal Hanif -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Owais Ahmad Sent: Tuesday, June 05, 2012 7:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] CDRs on multiple servers. Hello guys, I need to be able to throw cdrs on more than one servers at a time. Please let me know how this can be done. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDRs on multiple servers.
On Tue, 5 Jun 2012, Owais Ahmad wrote: I need to be able to throw cdrs on more than one servers at a time. Do you mean: 1) I need to collect the CDRs from multiple servers in one place so I can report on them. Or 2) I need to distribute the CDRs to multiple servers so I won't lose [m]any in case a host smokes. I like storing CDRs in a database (MySQL is my choice) so '1' is easy and '2' may be handled with database replication. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Cannot get Digium Phones back into service after changing sip device name.
During testing with the Digium phones I have run into a problem where I try to make a change to the sip device name. I make the device name change in sip.conf then make the matching change to the lines in res_digium_phone.conf. I then do 'sip reload' and 'module reload res_digium_phone.so'. I then end up with phones that I cannot bring into service no matter what I have tried. They act normally as far as seeing the configuration server, allowing me to enter the key, select the user, and then I receive the Error fetching config from proxy. message. I occasionally receive the following error in the CLI: NOTICE[5941]: phone_users.c:1626 token_set_last_info: Phone at '' reconfigureed. Another phone located at 'sip:10.72.65.114:5060' took over the config. I have tried Reset to Factory Defaults on the phones, I have tried clearing the keys from the Asterisk database, I have tried 'sip unregister', I have tried restarting Asterisk and rebooting. I cannot seem to get these test phones back into service. I am only in testing now where the phones are literally at arms reach and I am really nervous what can happen when we go into production. I really hope that I did something wrong in the process and that the phones are not really this fragile. My test system is currently running these versions: Asterisk 1.8.11-cert2 x86_32 DPMA Module: 1.8.11_1.0.1-x86_32 Digium Phone Firmware: 1_0_5_46476 Your help on this is really appreciated. Thank you. -- _ -- 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] Cannot get Digium Phones back into service after changing sip device name.
On 06/05/2012 10:23 AM, Chet W. Stevens wrote: During testing with the Digium phones I have run into a problem where I try to make a change to the sip device name. I make the device name change in sip.conf then make the matching change to the lines in res_digium_phone.conf. I then do 'sip reload' and 'module reload res_digium_phone.so'. I then end up with phones that I cannot bring into service no matter what I have tried. They act normally as far as seeing the configuration server, allowing me to enter the key, select the user, and then I receive the Error fetching config from proxy. message. I occasionally receive the following error in the CLI: NOTICE[5941]: phone_users.c:1626 token_set_last_info: Phone at '' reconfigureed. Another phone located at 'sip:10.72.65.114:5060' took over the config. I have tried Reset to Factory Defaults on the phones, I have tried clearing the keys from the Asterisk database, I have tried 'sip unregister', I have tried restarting Asterisk and rebooting. I cannot seem to get these test phones back into service. I am only in testing now where the phones are literally at arms reach and I am really nervous what can happen when we go into production. I really hope that I did something wrong in the process and that the phones are not really this fragile. My test system is currently running these versions: Asterisk 1.8.11-cert2 x86_32 DPMA Module: 1.8.11_1.0.1-x86_32 Digium Phone Firmware: 1_0_5_46476 Your help on this is really appreciated. Thank you. The first step would be to contact Digium technical support. They would be happy to assist you with any issues you're having. -- _ -- 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] Another IP address to block
Yesterday a customer was attacked from the following IP addresses so add them to your blacklist: iptables -A INPUT -s 37.8.119.75 -j DROP iptables -A INPUT -s 37.8.22.240 -j DROP -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- 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] Another IP address to block
We use complete regional blocks from Wizcraft and blocking at minimum all of unwanted Asia Pacific, Nigeria, Middle East, Russia, etc. We block almost anything that is not our actual customer market and screw the rest. On Tue, Jun 5, 2012 at 12:14 PM, Carlos Chavez cur...@telecomabmex.com wrote: Yesterday a customer was attacked from the following IP addresses so add them to your blacklist: iptables -A INPUT -s 37.8.119.75 -j DROP iptables -A INPUT -s 37.8.22.240 -j DROP -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 95, Issue 6
I figured out the problem. Actually the sending fax machine speed was set as 33000 bps, later I set to 14400 bps and in my dial plan, I forcefully set to use T.38 protocol. After that I was able to receive fax. Thanks Tim for assisting me out :). - Original Message - Hi Tim, I'm using Asterisk 10 and on Cisco GW the protocol is set for FAX is T.38 and when I try to send the fax from a fax machine i.e. HP 3180, I'm getting some warnings as listed below; -- Executing [4112345678@default:1] Goto(SIP/192.168.1.69-0005, fax-detect,fax,1) in new stack -- Goto (fax-detect,fax,1) -- Executing [fax@fax-detect:1] NoOp(SIP/192.168.1.69-0005, FAX DETECTED ) in new stack -- Executing [fax@fax-detect:2] Goto(SIP/192.168.1.69-0005, fax-receive,receive,1) in new stack -- Goto (fax-receive,receive,1) -- Executing [receive@fax-receive:1] NoOp(SIP/192.168.1.69-0005, FAX RECEIVE ) in new stack -- Executing [receive@fax-receive:2] Set(SIP/192.168.1.69-0005, GLOBAL(FAXCOUNT)=5) in new stack == Setting global variable 'FAXCOUNT' to '5' -- Executing [receive@fax-receive:3] Set(SIP/192.168.1.69-0005, FAXCOUNT=5) in new stack -- Executing [receive@fax-receive:4] Set(SIP/192.168.1.69-0005, FAXFILE=fax-5-rx.tif) in new stack -- Executing [receive@fax-receive:5] Set(SIP/192.168.1.69-0005, GLOBAL(LASTFAXCALLERNUM)=6461234567) in new stack == Setting global variable 'LASTFAXCALLERNUM' to '6461234567' -- Executing [receive@fax-receive:6] Set(SIP/192.168.1.69-0005, GLOBAL(LASTFAXCALLERNAME)=) in new stack == Setting global variable 'LASTFAXCALLERNAME' to '' -- Executing [receive@fax-receive:7] NoOp(SIP/192.168.1.69-0005, SETTING FAXOPT ) in new stack -- Executing [receive@fax-receive:8] Set(SIP/192.168.1.69-0005, FAXOPT(ecm)=yes) in new stack -- Executing [receive@fax-receive:9] Set(SIP/192.168.1.69-0005, FAXOPT(headerinfo)=MY FAXBACK RX) in new stack -- Executing [receive@fax-receive:10] Set(SIP/192.168.1.69-0005, FAXOPT(localstationid)=1234567890) in new stack -- Executing [receive@fax-receive:11] Set(SIP/192.168.1.69-0005, FAXOPT(maxrate)=14400) in new stack -- Executing [receive@fax-receive:12] Set(SIP/192.168.1.69-0005, FAXOPT(minrate)=2400) in new stack -- Executing [receive@fax-receive:13] NoOp(SIP/192.168.1.69-0005, FAXOPT(ecm) : yes) in new stack -- Executing [receive@fax-receive:14] NoOp(SIP/192.168.1.69-0005, FAXOPT(headerinfo) : MY FAXBACK RX) in new stack -- Executing [receive@fax-receive:15] NoOp(SIP/192.168.1.69-0005, FAXOPT(localstationid) : 1234567890) in new stack -- Executing [receive@fax-receive:16] NoOp(SIP/192.168.1.69-0005, FAXOPT(maxrate) : 14400) in new stack -- Executing [receive@fax-receive:17] NoOp(SIP/192.168.1.69-0005, FAXOPT(minrate) : 2400) in new stack -- Executing [receive@fax-receive:18] NoOp(SIP/192.168.1.69-0005, RECEIVING FAX : fax-5-rx.tif ) in new stack -- Executing [receive@fax-receive:19] ReceiveFAX(SIP/192.168.1.69-0005, /var/spool/asterisk/fax/fax-5-rx.tif) in new stack -- Channel 'SIP/192.168.1.69-0005' receiving FAX '/var/spool/asterisk/fax/fax-5-rx.tif' == Using UDPTL CoS mark 5 [Jun 4 12:35:02] NOTICE[10371]: chan_sip.c:7577 sip_read: FAX CNG detected but no fax extension [Jun 4 12:35:02] WARNING[10072]: res_fax.c:1666 receivefax_t38_init: channel 'SIP/192.168.1.69-0005' refused to negotiate T.38 [Jun 4 12:35:02] WARNING[10072]: res_fax.c:1687 receivefax_t38_init: Audio FAX not allowed on channel 'SIP/192.168.1.69-0005' and T.38 negotiation failed; aborting. [Jun 4 12:35:02] ERROR[10072]: res_fax.c:1891 receivefax_exec: error initializing channel 'SIP/192.168.1.69-0005' in T.38 mode == Spawn extension (fax-receive, receive, 19) exited non-zero on 'SIP/192.168.1.69-0005' In my sip.conf global configuration I enabled 'fax detect' and 't38pt_udptl' and added Cisco VGW peer; [CiscoVGW-10.70.X.X] host=10.70.X.X type=friend disallow=all allow=ulaw allow=alaw nat=yes insecure=port,invite context=fax-call canreinvite=no qualify=yes dtmfmode=inband T.38 failed to negotiate. That means either your Asterisk side, or your Cisco side are not playing nicely together. A packet capture of the call setup would be helpful to determine which side is having the issues. --Tim -- Regards, Ahmed Munir Chohan -- _ -- 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] Fax over IP ?
On Tue, Jun 5, 2012 at 12:47 PM, Ahmed Munir ahmedmunir...@gmail.comwrote: I figured out the problem. Actually the sending fax machine speed was set as 33000 bps, later I set to 14400 bps and in my dial plan, I forcefully set to use T.38 protocol. After that I was able to receive fax. Thanks Tim for assisting me out :). - Original Message - Hi Tim, I'm using Asterisk 10 and on Cisco GW the protocol is set for FAX is T.38 and when I try to send the fax from a fax machine i.e. HP 3180, I'm getting some warnings as listed below; -- Executing [4112345678@default:1] Goto(SIP/192.168.1.69-0005, fax-detect,fax,1) in new stack -- Goto (fax-detect,fax,1) -- Executing [fax@fax-detect:1] NoOp(SIP/192.168.1.69-0005, FAX DETECTED ) in new stack -- Executing [fax@fax-detect:2] Goto(SIP/192.168.1.69-0005, fax-receive,receive,1) in new stack -- Goto (fax-receive,receive,1) -- Executing [receive@fax-receive:1] NoOp(SIP/192.168.1.69-0005, FAX RECEIVE ) in new stack -- Executing [receive@fax-receive:2] Set(SIP/192.168.1.69-0005, GLOBAL(FAXCOUNT)=5) in new stack == Setting global variable 'FAXCOUNT' to '5' -- Executing [receive@fax-receive:3] Set(SIP/192.168.1.69-0005, FAXCOUNT=5) in new stack -- Executing [receive@fax-receive:4] Set(SIP/192.168.1.69-0005, FAXFILE=fax-5-rx.tif) in new stack -- Executing [receive@fax-receive:5] Set(SIP/192.168.1.69-0005, GLOBAL(LASTFAXCALLERNUM)=6461234567) in new stack == Setting global variable 'LASTFAXCALLERNUM' to '6461234567' -- Executing [receive@fax-receive:6] Set(SIP/192.168.1.69-0005, GLOBAL(LASTFAXCALLERNAME)=) in new stack == Setting global variable 'LASTFAXCALLERNAME' to '' -- Executing [receive@fax-receive:7] NoOp(SIP/192.168.1.69-0005, SETTING FAXOPT ) in new stack -- Executing [receive@fax-receive:8] Set(SIP/192.168.1.69-0005, FAXOPT(ecm)=yes) in new stack -- Executing [receive@fax-receive:9] Set(SIP/192.168.1.69-0005, FAXOPT(headerinfo)=MY FAXBACK RX) in new stack -- Executing [receive@fax-receive:10] Set(SIP/192.168.1.69-0005, FAXOPT(localstationid)=1234567890) in new stack -- Executing [receive@fax-receive:11] Set(SIP/192.168.1.69-0005, FAXOPT(maxrate)=14400) in new stack -- Executing [receive@fax-receive:12] Set(SIP/192.168.1.69-0005, FAXOPT(minrate)=2400) in new stack -- Executing [receive@fax-receive:13] NoOp(SIP/192.168.1.69-0005, FAXOPT(ecm) : yes) in new stack -- Executing [receive@fax-receive:14] NoOp(SIP/192.168.1.69-0005, FAXOPT(headerinfo) : MY FAXBACK RX) in new stack -- Executing [receive@fax-receive:15] NoOp(SIP/192.168.1.69-0005, FAXOPT(localstationid) : 1234567890) in new stack -- Executing [receive@fax-receive:16] NoOp(SIP/192.168.1.69-0005, FAXOPT(maxrate) : 14400) in new stack -- Executing [receive@fax-receive:17] NoOp(SIP/192.168.1.69-0005, FAXOPT(minrate) : 2400) in new stack -- Executing [receive@fax-receive:18] NoOp(SIP/192.168.1.69-0005, RECEIVING FAX : fax-5-rx.tif ) in new stack -- Executing [receive@fax-receive:19] ReceiveFAX(SIP/192.168.1.69-0005, /var/spool/asterisk/fax/fax-5-rx.tif) in new stack -- Channel 'SIP/192.168.1.69-0005' receiving FAX '/var/spool/asterisk/fax/fax-5-rx.tif' == Using UDPTL CoS mark 5 [Jun 4 12:35:02] NOTICE[10371]: chan_sip.c:7577 sip_read: FAX CNG detected but no fax extension [Jun 4 12:35:02] WARNING[10072]: res_fax.c:1666 receivefax_t38_init: channel 'SIP/192.168.1.69-0005' refused to negotiate T.38 [Jun 4 12:35:02] WARNING[10072]: res_fax.c:1687 receivefax_t38_init: Audio FAX not allowed on channel 'SIP/192.168.1.69-0005' and T.38 negotiation failed; aborting. [Jun 4 12:35:02] ERROR[10072]: res_fax.c:1891 receivefax_exec: error initializing channel 'SIP/192.168.1.69-0005' in T.38 mode == Spawn extension (fax-receive, receive, 19) exited non-zero on 'SIP/192.168.1.69-0005' In my sip.conf global configuration I enabled 'fax detect' and 't38pt_udptl' and added Cisco VGW peer; [CiscoVGW-10.70.X.X] host=10.70.X.X type=friend disallow=all allow=ulaw allow=alaw nat=yes insecure=port,invite context=fax-call canreinvite=no qualify=yes dtmfmode=inband T.38 failed to negotiate. That means either your Asterisk side, or your Cisco side are not playing nicely together. A packet capture of the call setup would be helpful to determine which side is having the issues. --Tim -- Regards, Ahmed Munir Chohan -- Regards, Ahmed Munir Chohan -- _ -- 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:
[asterisk-users] G729 and voice mail
I am trying to figure out the best way to deal with this. I want all of the calls in the network to be G729 and this is working. I do have hardware that provides me 30 g729 licenses. I am setting each extensions to disallow=all and allow=g729. However when I have this setup, I get no voice mail prompts. I tried setting to disallow=all and allow=g729,alaw and I still have no audio when calling voice mail. If I remove the disallow=all I do have voice mail prompts, but the calls do not seem to be always using g729 when possible. -- _ -- 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] G729 and voice mail
What does the output of g729 show licenses show? If it doesn't show licenses then Asterisk is not licensed for G729 codec. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim King Sent: Tuesday, June 05, 2012 2:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] G729 and voice mail I am trying to figure out the best way to deal with this. I want all of the calls in the network to be G729 and this is working. I do have hardware that provides me 30 g729 licenses. I am setting each extensions to disallow=all and allow=g729. However when I have this setup, I get no voice mail prompts. I tried setting to disallow=all and allow=g729,alaw and I still have no audio when calling voice mail. If I remove the disallow=all I do have voice mail prompts, but the calls do not seem to be always using g729 when possible. -- _ -- 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] G729 and voice mail
The G729 is coming from a Sangoma D100-030 card and the G729 transcoding is working, the only issue I have is not hearing prompts from the system. On Tue, Jun 5, 2012 at 2:42 PM, Eric Wieling ewiel...@nyigc.com wrote: What does the output of g729 show licenses show? If it doesn't show licenses then Asterisk is not licensed for G729 codec. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Tim King Sent: Tuesday, June 05, 2012 2:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] G729 and voice mail I am trying to figure out the best way to deal with this. I want all of the calls in the network to be G729 and this is working. I do have hardware that provides me 30 g729 licenses. I am setting each extensions to disallow=all and allow=g729. However when I have this setup, I get no voice mail prompts. I tried setting to disallow=all and allow=g729,alaw and I still have no audio when calling voice mail. If I remove the disallow=all I do have voice mail prompts, but the calls do not seem to be always using g729 when possible. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 and voice mail
When you installed asterisk, did you also enable the installation of the g729 asterisk-(foo)-sounds options in 'make menuconfig'? On 6/5/2012 2:46 PM, Tim King wrote: The G729 is coming from a Sangoma D100-030 card and the G729 transcoding is working, the only issue I have is not hearing prompts from the system. On Tue, Jun 5, 2012 at 2:42 PM, Eric Wieling ewiel...@nyigc.com mailto:ewiel...@nyigc.com wrote: What does the output of g729 show licenses show? If it doesn't show licenses then Asterisk is not licensed for G729 codec. -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim King Sent: Tuesday, June 05, 2012 2:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] G729 and voice mail I am trying to figure out the best way to deal with this. I want all of the calls in the network to be G729 and this is working. I do have hardware that provides me 30 g729 licenses. I am setting each extensions to disallow=all and allow=g729. However when I have this setup, I get no voice mail prompts. I tried setting to disallow=all and allow=g729,alaw and I still have no audio when calling voice mail. If I remove the disallow=all I do have voice mail prompts, but the calls do not seem to be always using g729 when possible. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] unable to create channel of type 'SIP'
Can you give me some pointers on where to read documentation on how to
set up registered phones?
Also I'm wondering if maybe it would help if I tried setting up some
softphones first.
Can someone recommend some cheap softphones that work with asterisk?
Jacob
On Tue, May 29, 2012 at 5:36 PM, Danny Nicholas da...@debsinc.com wrote:
You can dial out from an unregistered SIP peer, but you can't receive a call
or call that peer.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jacob Fenwick
Sent: Tuesday, May 29, 2012 4:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] unable to create channel of type 'SIP'
Good catch.
Unfortunately, I actually did have it in there as dialGSM, I just copied
from the wrong version of the file when I copied and pasted it here.
This is what I get from sip show peers:
Name/Username: IMSI262422146099205
Host: (Unspecified)
Dyn: D
Forceport: 0
ACL:
Port: Unmonitored
Status
... same for the other IMSI...
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2
offline]
Jacob
On Tue, May 29, 2012 at 5:25 PM, James Thomas jthomas...@gmail.com wrote:
I think you need to change:
exten = 2012,1,Macro(dialSIP,IMSI262428511722625)
exten = 2013,1,Macro(dialSIP,IMSI262422146099205)
to:
exten = 2012,1,Macro(dialGSM,IMSI262428511722625)
exten = 2013,1,Macro(dialGSM,IMSI262422146099205)
also what does sip show peers show, as opposed to sip show registry?
On Tue, May 29, 2012 at 2:55 PM, Jacob Fenwick
jacob.fenw...@gmail.com
wrote:
I'm trying to use OpenBTS with Asterisk.
I have two phones that are connecting to OpenBTS correctly, but on
the Asterisk side the phones can't call each other.
I followed this guide:
http://gnuradio.org/redmine/projects/gnuradio/wiki/OpenBTSSettingUpAs
terisk I set up two phones in sip.conf and extensions.conf.
In my SIP output I see this:
WARNING[1689]: app_dial.c:2041 dial_exec_full: unable to create
channel of type 'SIP' (cause 20 - unknown)
If I type sip show registry it says there are 0 SIP registrations.
Should I see both the phones registered at this point?
If that's what's wrong, what am I doing wrong that's making the
phones not able to register?
Below is my Asterisk configuration.
Jacob
#/etc/asterisk/sip.conf
[general]
context=sip-external
#...
[IMSI262428511722625]
callerid=2012
canreinvite=no
type=friend
context=sip-external
allow=gsm
host=dynamic
dtmfmode=info
[IMSI262422146099205]
callerid=2013
canreinvite=no
type=friend
context=sip-external
allow=gsm
host=dynamic
dtmfmode=info
#/etc/asterisk/extensions.conf
[macro-dialGSM]
exten = s,1,Dial(SIP/${ARG1})
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-CANCEL,1,Hangup
exten = s-NOANSWER,1,Hangup
exten = s-BUSY,1,Busy(30)
exten = s-CONGESTION,1,Congestion(30) exten =
s-CHANUNAVAIL,1,playback(ss-noservice)
exten = s-CANCEL,1,Hangup
[sip-external]
exten = 2012,1,Macro(dialSIP,IMSI262428511722625)
exten = 2013,1,Macro(dialSIP,IMSI262422146099205)
--
_
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--
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Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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asterisk-users
[asterisk-users] Hylafax + OpenVz + Iptables
Hello, I know this a little bit off topic but maybe someone of you has experience with this topic. I want to run a hylafax install in an openvz container with iptables. My Problem is that I can load the kernel module for ftp connection tracking (nf_conntrack_ftp) in the containers config file but no idea how to set the parameter to track the right ports. My vz config looks like: IPTABLES=ipt_REJECT ipt_tos ipt_TOS ipt_LOG ip_conntrack ipt_limit ipt_multiport iptable_filter iptable_mangle ipt_TCPMSS ipt_tcpmss ipt_ttl ipt_length ipt_state iptable_nat ip_nat_ftp ip_conntrack_ftp ipt_conntrack ip_tables On the host machine the module nf_conntrack_ftp ports is loaded with the parameter ports=21,4559. With best regards David Oppermann Voip Engineer // v...@sil.at // Tel 059944-2440 // - SILVER SERVER GmbH - a Tele2 Company // Donau-City-Strasse 11 // A-1220 Wien // Fax 059944-9000 // www.sil.at // FN 204414i // Handelsgericht Wien // UID ATU 51064903 // - -- _ -- 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
