[asterisk-users] Diva Server BRI hangs up after about 25 seconds
I have a Diva Server BRI installed in a Debian system running Asterisk 1.2.22. When a call comes in the Background application plays a couple of times. Halfway through the second time, the call is disconnected (total time connected about 25 seconds). I've posted various configs below. Any advice on how to debug would be appreciated. We are located in New Zealand. CLI showing capi debug (just before hangup) -- EICONISDN#02: DATA_B3_IND (len=160) fr.datalen=160 fr.subclass=8 DATA_B3_REQ ID=001 #0x04e1 LEN=0030 Controller/PLCI/NCCI= 0x10201 Data32 = 0x816d494 DataLength = 0xa0 DataHandle = 0x4d9 Flags = 0x0 Data64 = 0x0 DATA_B3_CONF ID=001 #0x04e1 LEN=0016 Controller/PLCI/NCCI= 0x10201 DataHandle = 0x4d9 Info= 0x0 DATA_B3_IND ID=001 #0x04ed LEN=0022 Controller/PLCI/NCCI= 0x10201 Data32 = 0x405b1362 DataLength = 0xa0 DataHandle = 0x163 Flags = 0x0 Data64 = 0x8b8b8b8b8b8b8b8b DATA_B3_RESP ID=001 #0x04ed LEN=0014 Controller/PLCI/NCCI= 0x10201 DataHandle = 0x163 -- EICONISDN#02: DATA_B3_IND (len=160) fr.datalen=160 fr.subclass=8 DATA_B3_REQ ID=001 #0x04e2 LEN=0030 Controller/PLCI/NCCI= 0x10201 Data32 = 0x816d574 DataLength = 0xa0 DataHandle = 0x4da Flags = 0x0 Data64 = 0x0 DATA_B3_CONF ID=001 #0x04e2 LEN=0016 Controller/PLCI/NCCI= 0x10201 DataHandle = 0x4da Info= 0x0 INFO_IND ID=001 #0x04ee LEN=0017 Controller/PLCI/NCCI= 0x201 InfoNumber = 0x8 InfoElement = 82 90 INFO_RESP ID=001 #0x04ee LEN=0012 Controller/PLCI/NCCI= 0x201 -- EICONISDN#02: info element CAUSE 82 90 INFO_IND ID=001 #0x04ef LEN=0015 Controller/PLCI/NCCI= 0x201 InfoNumber = 0x8045 InfoElement = default INFO_RESP ID=001 #0x04ef LEN=0012 Controller/PLCI/NCCI= 0x201 -- EICONISDN#02: info element DISCONNECT -- EICONISDN#02: Disconnect case 3 -- CAPI queue frame:[ TYPE: Control (4) SUBCLASS: Hangup (1) ] [EICONISDN#02] == Spawn extension (incoming, 09375, 3) exited non-zero on 'CAPI/EICONISDN/09375-0' == EICONISDN#02: CAPI Hangingup for PLCI=0x201 in state 2 -- EICONISDN#02: activehangingup (cause=16) for PLCI=0x201 DISCONNECT_B3_REQ ID=001 #0x04e3 LEN=0013 Controller/PLCI/NCCI= 0x10201 NCPI= default DISCONNECT_B3_CONF ID=001 #0x04e3 LEN=0014 Controller/PLCI/NCCI= 0x10201 Info= 0x0 DISCONNECT_B3_IND ID=001 #0x04f1 LEN=0015 Controller/PLCI/NCCI= 0x10201 Reason_B3 = 0x0 NCPI= default DISCONNECT_B3_RESP ID=001 #0x04f1 LEN=0012 Controller/PLCI/NCCI= 0x10201 DISCONNECT_REQ ID=001 #0x04e4 LEN=0018 Controller/PLCI/NCCI= 0x201 AdditionalInfo BChannelinformation= default Keypadfacility = default Useruserdata = default Facilitydataarray = default SendingComplete= default CAPI devicestate requested for EICONISDN/09375 CAPI devicestate requested for EICONISDN/09375 DISCONNECT_CONF ID=001 #0x04e4 LEN=0014 Controller/PLCI/NCCI= 0x201 Info= 0x0 INFO_IND ID=001 #0x04f2 LEN=0015 Controller/PLCI/NCCI= 0x201 InfoNumber = 0x805a InfoElement = default INFO_RESP ID=001 #0x04f2 LEN=0012 Controller/PLCI/NCCI= 0x201 -- EICONISDN#02: info element RELEASE COMPLETE DISCONNECT_IND ID=001 #0x04f4 LEN=0014 Controller/PLCI/NCCI= 0x201 Reason = 0x3490 DISCONNECT_RESP ID=001 #0x04f4 LEN=0012 Controller/PLCI/NCCI= 0x201 EICONISDN#02: CAPI INFO 0x3490: Normal call clearing == EICONISDN#02: Interface cleanup PLCI=0x201 /usr/lib/divas/Config Interface mode: TE (verified) D Channel: ETSI-DSS1 (verified) NT-2 mode: No D-Channel Layer activation: Deactivation by other side Voice companding: National default Hunt group operation: Standard Trunk Operation mode: point to point (fixed TEI) (verified) TEI value: 0 (verified) Source of tones: Provided by ISDN CAPI call distribution: Off Max fax speed: No limit Min fax speed: No limit Fax session limit: 0 T.30 protocol options: None selected Part 68 voice signal
[asterisk-users] ZT_CHANCONFIG failed on channel 1: No such device or address
I have had a TDM400 with 2 FXO and 2 FXS working for ages (12 months). It has stopped working. All four green lights are still lit. I have rebuilt zaptel and asterisk and restarted but the problem persists. /sbin/ztcfg - Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) ZT_CHANCONFIG failed on channel 1: No such device or address vi /etc/zaptel.conf fxoks=1 fxoks=2 fxsks=3 fxsks=4 loadzone=nz defaultzone=nz lsmod | grep zaptel zaptel183076 2 zttranscode,wctdm crc_ccitt 6465 1 zaptel lspci Card is not listed dmesg Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.2.16 Zaptel Echo Canceller: KB1 Zaptel Transcoder support loaded vi /etc/udev/rules.d/50-udev.rules # Section for zaptel device KERNEL==zapctl, NAME=zap/ctl KERNEL==zaptimer, NAME=zap/timer KERNEL==zapchannel, NAME=zap/channel KERNEL==zappseudo,NAME=zap/pseudo KERNEL==zap[0-9]*,NAME=zap/%n vi /etc/modprobe.conf install wctdm /sbin/modprobe --ignore-install wctdm opermode=NEWZEALAND fxshonormode=1 boostringer=1 fastringer=1 /sbin/ztcfg ls /proc/zaptel ls -la /dev/zap drwxr-xr-x 2 asterisk asterisk 140 Apr 19 20:05 . drwxr-xr-x 11 root root 3620 Apr 19 21:01 .. crw--- 1 asterisk asterisk 196, 254 Apr 19 20:05 channel crw--- 1 asterisk asterisk 196, 0 Apr 19 20:05 ctl crw--- 1 asterisk asterisk 196, 255 Apr 19 20:05 pseudo crw--- 1 asterisk asterisk 196, 253 Apr 19 20:05 timer crw-rw 1 asterisk asterisk 196, 250 Apr 19 20:05 transcode Does all that meant that Linux can't see the card? Any suggestions greatly appreciated. uname -r 2.6.18-1.2257.fc5smp Asterisk 1.2.17 Zaptel 1.2.16 Regards Cameron ___ Yahoo! Mail is the world's favourite email. Don't settle for less, sign up for your free account today http://uk.rd.yahoo.com/evt=44106/*http://uk.docs.yahoo.com/mail/winter07.html ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZT_CHANCONFIG failed on channel 1: No such device or address
What is the output of: ls -l /sys/class/zaptel ls -l /sys/class/zaptel total 0 drwxr-xr-x 2 root root 0 Apr 19 20:05 zapchannel drwxr-xr-x 2 root root 0 Apr 19 20:05 zapctl drwxr-xr-x 2 root root 0 Apr 19 20:05 zappseudo drwxr-xr-x 2 root root 0 Apr 19 20:05 zaptimer drwxr-xr-x 2 root root 0 Apr 19 20:05 zaptranscode ___ Yahoo! Mail is the world's favourite email. Don't settle for less, sign up for your free account today http://uk.rd.yahoo.com/evt=44106/*http://uk.docs.yahoo.com/mail/winter07.html ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] make Zaptel 1.2.16 'struct inode' has no member named 'u'.
Zaptel has no direct code relationship with Asterisk. Your error is because zaptel is trying to use a member no longer exists in newer kernels. However you are using fedora, and fedora included that change in older kernel. I found this in xpp/xbus-core.c /* * As part of the inode diet the private data member of struct inode * has changed in 2.6.19. However, Fedore Core 6 adopted this change * a bit earlier (2.6.18). If you use such a kernel, Change the * following test from 2,6,19 to 2,6,18. */ #if LINUX_VERSION_CODE KERNEL_VERSION(2,6,19) #define I_PRIVATE(inode)((inode)-u.generic_ip) #else #define I_PRIVATE(inode)((inode)-i_private) #endif The following resolved this issue: vi xpp/xbus-core.c Change code as follows: #if LINUX_VERSION_CODE KERNEL_VERSION(2,6,18) make clean make Thanks Cameron ___ What kind of emailer are you? Find out today - get a free analysis of your email personality. Take the quiz at the Yahoo! Mail Championship. http://uk.rd.yahoo.com/evt=44106/*http://mail.yahoo.net/uk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] make Zaptel 1.2.16 'struct inode' has no member named 'u'.
On attempting to make Zaptel 1.2.16 on FC5, I get the following messages: /usr/src/zaptel-1.2.16/xpp/xbus-core.c: In function 'debugfs_open': /usr/src/zaptel-1.2.16/xpp/xbus-core.c:171: error: 'struct inode' has no member named 'u' make[3]: *** [/usr/src/zaptel-1.2.16/xpp/xbus-core.o] Error 1 make[2]: *** [/usr/src/zaptel-1.2.16/xpp] Error 2 make[1]: *** [_module_/usr/src/zaptel-1.2.16] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.18-1.2257.fc5-smp-i686' make: *** [all] Error 2 An internet search has turned this message up but other than indicating that the inode structure has changed I'm no further ahead. I have found nothing specific for Asterisk. Any advice appreciated. Cameron ___ The all-new Yahoo! Mail goes wherever you go - free your email address from your Internet provider. http://uk.docs.yahoo.com/nowyoucan.html ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SDP bug
The call that gets dropped had a retransmission of INVITE from UAC to UAS (and therefore retransmission of 200 OK from UAS to UAC). There is nothing wrong with the re-transmission as such, but I noticed a potential bug in Asterisk in the way it responds to an INVITE retransmission. Asterisk is bumping up the session version number in the retransmitted 200 OK's SDP. This is as if Asterisk is treating the INVITE retransmission as a RE-INVITE. Asterisk sends 200 OK: o=root 16300 16300 IN IP4 203.89.nnn.nnn Asterisk sends 200 OK (retransmission): o=root 16300 16301 IN IP4 203.89.nnn.nnn Ideally, this bug should have nothing to do with why Asterisk is ignoring the ACK (which is why it keeps reatrasmitting the 200 OK and eventually drops the call). However, if you can confirm that all dropped calls have INVITE retransmission then that might give us a clue? Raj, That's an interesting observation. Do you think this will cause any issues? Even though it's not beautiful, I fail to see why a UA would check that. I have run a number of tests and in all cases the calls that fail have a retransmitted INVITE whereas the successfull calls have only one INVITE. Regards Cameron ___ Now you can scan emails quickly with a reading pane. Get the new Yahoo! Mail. http://uk.docs.yahoo.com/nowyoucan.html ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk hangs up SIP call after 6 200 retransmits
One potential reason could be that the ACK request being sent to Asterisk is malformed. Notice branch=0 in the top Via. This should start with z9hG4bK magic cookie since the INVITE was an RFC 3261 transaction. While branch=0 is valid in RFC 2543, I don't think an INVITE can start-off as RFC 3261 and then the ACK can switch over to RFC 2543 in the middle of the transaction. Clearly, Asterisk is dropping this ACK on the floor. OK. But in the calls that don't get dropped, the branch=0 is present also. See below for an example: -- SIP read from 147.202.nnn.nnn:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Record-Route: sip:147.202.nnn.nnn;ftag=as1370b1ab;lr=on Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bK22ab.697375a4.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK4cf2bb78;rport=5060 From: 649444 sip:[EMAIL PROTECTED];tag=as1370b1ab To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Mon, 02 Apr 2007 03:37:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 338 v=0 o=root 11402 11402 IN IP4 202.180.nnn.nnn s=session c=IN IP4 202.180.nnn.nnn t=0 0 m=audio 39686 RTP/AVP 18 97 3 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (15 headers 15 lines) --- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 147.202.nnn.nnn : 5060 (non-NAT) Found peer 'DLS' Found RTP audio format 18 Found RTP audio format 97 Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 202.180.nnn.nnn:39686 Found description format G729 Found description format iLBC Found description format GSM Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x50e (gsm|ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 649977 in from-trunk (domain 203.89.nnn.nnn) list_route: hop: sip:147.202.nnn.nnn;ftag=as1370b1ab;lr=on Transmitting (no NAT) to 147.202.nnn.nnn:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bK22ab.697375a4.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK4cf2bb78;rport=5060 From: 649444 sip:[EMAIL PROTECTED];tag=as1370b1ab To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- -- Goto (ivr-3,s,1) -- Executing Set(SIP/649977-b7908550, LOOPCOUNT=0) in new stack -- Executing Set(SIP/649977-b7908550, __DIR-CONTEXT=11000111000) in new stack -- Executing Answer(SIP/649977-b7908550, ) in new stack We're at 203.89.nnn.nnn port 15804 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 147.202.nnn.nnn:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bK22ab.697375a4.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK4cf2bb78;rport=5060 Record-Route: sip:147.202.nnn.nnn;ftag=as1370b1ab;lr=on From: 649444 sip:[EMAIL PROTECTED];tag=as1370b1ab To: sip:[EMAIL PROTECTED];tag=as7ecf44d1 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 15804 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Executing Wait(SIP/649977-b7908550, 1) in new stack capetown*CLI -- SIP read from 147.202.nnn.nnn:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Record-Route: sip:147.202.nnn.nnn;ftag=as1370b1ab;lr=on Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK5ba4f251;rport=5060 From: 649444 sip:[EMAIL PROTECTED];tag=as1370b1ab To: sip:[EMAIL PROTECTED];tag=as7ecf44d1 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 --- (12 headers 0 lines) --- -- Executing Set(SIP/649977-b7908550, TIMEOUT(digit)=3) in new stack -- Digit timeout set to 3 -- Executing Set(SIP/649977-b7908550, TIMEOUT(response)=10) in new stack -- Response timeout set to 10 -- Executing BackGround(SIP/649977-b7908550, custom/11000111000-welcome)
[asterisk-users] Asterisk hangs up SIP call after 6 200 retransmits
I have the following scenario: PSTN gateway (202.180.nnn.nnn) - OpenSER 1.0.1 (147.202.nnn.nnn) - Asterisk 1.2.16 (203.89.nnn.nnn) When an incoming call is received, often (but not always) Asterisk repeatedly sends a SIP 200 OK message and eventually hangs up the call. sip.conf [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) externip = 203.89.nnn.nnn disallow=all allow=ulaw allow=alaw language=nz [DLS] username=649977 type=peer host=domain.co.nz context=from-trunk canreinvite=no Note that Asterisk registers with proxy: 649977:[EMAIL PROTECTED]/649977 sip debug peer DLS -- SIP read from 147.202.nnn.nnn:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Record-Route: sip:147.202.nnn.nnn;ftag=as4917b107;lr=on Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd0e3.48331fd3.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK3757e55c;rport=5060 From: 649444 sip:[EMAIL PROTECTED];tag=as4917b107 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Thu, 29 Mar 2007 17:00:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 338 v=0 o=root 13636 13636 IN IP4 202.180.nnn.nnn s=session c=IN IP4 202.180.nnn.nnn t=0 0 m=audio 36274 RTP/AVP 18 97 3 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (15 headers 15 lines) --- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 147.202.nnn.nnn : 5060 (non-NAT) Found peer 'DLS' Found RTP audio format 18 Found RTP audio format 97 Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 202.180.nnn.nnn:36274 Found description format G729 Found description format iLBC Found description format GSM Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x50e (gsm|ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 649977 in from-trunk (domain 203.89.nnn.nnn) list_route: hop: sip:147.202.nnn.nnn;ftag=as4917b107;lr=on Transmitting (no NAT) to 147.202.nnn.nnn:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd0e3.48331fd3.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK3757e55c;rport=5060 From: 649444 sip:[EMAIL PROTECTED];tag=as4917b107 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- -- Executing Goto(SIP/649977-b791bb60, ivr-3|s|1) in new stack -- Goto (ivr-3,s,1) -- Executing Set(SIP/649977-b791bb60, LOOPCOUNT=0) in new stack -- Executing Set(SIP/649977-b791bb60, __DIR-CONTEXT=11000111000) in new stack -- Executing Answer(SIP/649977-b791bb60, ) in new stack We're at 203.89.nnn.nnn port 11648 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 147.202.nnn.nnn:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd0e3.48331fd3.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK3757e55c;rport=5060 Record-Route: sip:147.202.nnn.nnn;ftag=as4917b107;lr=on From: 649444 sip:[EMAIL PROTECTED];tag=as4917b107 To: sip:[EMAIL PROTECTED];tag=as7cefaa53 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 11648 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Executing Wait(SIP/649977-b791bb60, 1) in new stack capetown*CLI -- SIP read from 147.202.nnn.nnn:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Record-Route: sip:147.202.nnn.nnn;ftag=as4917b107;lr=on Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd0e3.48331fd3.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK3757e55c;rport=5060 From: 649444 sip:[EMAIL PROTECTED];tag=as4917b107 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Thu, 29 Mar 2007 17:00:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type:
[asterisk-users] Asterisk hangs up SIP call after 6 200 retransmits
I have the following scenario: PSTN gateway (202.180.nnn.nnn) - OpenSER 1.0.1 (147.202.nnn.nnn) - Asterisk 1.2.16 (203.89.nnn.nnn) When an incoming call is received, often (but not always) Asterisk repeatedly sends a SIP 200 OK message and eventually hangs up the call. sip.conf [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) externip = 203.89.nnn.nnn disallow=all allow=ulaw allow=alaw language=nz [DLS] username=649977 type=peer host=domain.co.nz context=from-trunk canreinvite=no Note that Asterisk registers with proxy: 649977:[EMAIL PROTECTED]/649977 sip debug peer DLS -- SIP read from 147.202.nnn.nnn:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Record-Route: sip:147.202.nnn.nnn;ftag=as4917b107;lr=on Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd0e3.48331fd3.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK3757e55c;rport=5060 From: 649444 sip:[EMAIL PROTECTED];tag=as4917b107 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Thu, 29 Mar 2007 17:00:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 338 v=0 o=root 13636 13636 IN IP4 202.180.nnn.nnn s=session c=IN IP4 202.180.nnn.nnn t=0 0 m=audio 36274 RTP/AVP 18 97 3 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (15 headers 15 lines) --- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 147.202.nnn.nnn : 5060 (non-NAT) Found peer 'DLS' Found RTP audio format 18 Found RTP audio format 97 Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 202.180.nnn.nnn:36274 Found description format G729 Found description format iLBC Found description format GSM Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x50e (gsm|ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 649977 in from-trunk (domain 203.89.nnn.nnn) list_route: hop: sip:147.202.nnn.nnn;ftag=as4917b107;lr=on Transmitting (no NAT) to 147.202.nnn.nnn:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd0e3.48331fd3.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK3757e55c;rport=5060 From: 649444 sip:[EMAIL PROTECTED];tag=as4917b107 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- -- Executing Goto(SIP/649977-b791bb60, ivr-3|s|1) in new stack -- Goto (ivr-3,s,1) -- Executing Set(SIP/649977-b791bb60, LOOPCOUNT=0) in new stack -- Executing Set(SIP/649977-b791bb60, __DIR-CONTEXT=11000111000) in new stack -- Executing Answer(SIP/649977-b791bb60, ) in new stack We're at 203.89.nnn.nnn port 11648 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 147.202.nnn.nnn:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd0e3.48331fd3.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK3757e55c;rport=5060 Record-Route: sip:147.202.nnn.nnn;ftag=as4917b107;lr=on From: 649444 sip:[EMAIL PROTECTED];tag=as4917b107 To: sip:[EMAIL PROTECTED];tag=as7cefaa53 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 11648 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Executing Wait(SIP/649977-b791bb60, 1) in new stack capetown*CLI -- SIP read from 147.202.nnn.nnn:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Record-Route: sip:147.202.nnn.nnn;ftag=as4917b107;lr=on Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd0e3.48331fd3.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK3757e55c;rport=5060 From: 649444 sip:[EMAIL PROTECTED];tag=as4917b107 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Thu, 29 Mar 2007 17:00:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp
Re: [asterisk-users] FW: Seamless Multi Office Asterisk Deployment
The issue with FreePBX is that it uses the Asterisk database to store user and device information (e.g. who is the currently logged-in user). So you need to replicate that information across multiple machines. The approach we have taken is to customise FreePBX (not trivial) so that all this information is stored (and looked up) in MySQL. In addition we store the context information to enable partitioning of the dialplan. Then use MySQL replication to push the values out to multiple servers. You could use this method to enable Roaming Extensions. You would need a script to push any configuration changes (since FreePBX stores config in the standard flat files) out to the various Asterisk servers (maybe using rsync) and reload the config. Alternatively you could use NFS and store the config centrally (reload still required). Regarding voicemail and recordings, you could use the same approach. We don't use Branch Unification (yet). You may wish to consider OpenSER as the registrar and then farming out to the various Asterisk servers as appropriate. Hope that gives you some ideas. Cameron - Original Message From: Brandon Comouche [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, 13 March, 2007 6:11:12 AM Subject: [asterisk-users] FW: Seamless Multi Office Asterisk Deployment Hello I have a brief and a long question about a possible Asterisk deployment I am planning. Long Story Short: I have four total offices, one main and three remote. All offices are connected using dedicated network T1 lines creating one unified network across offices. I would like to know if it is possible to set up an Asterisk system with the following capabilities: - Branch Unification (I know this can be done) - Branch Independence (In case of T1 network Failure, PSTN line failover at each branch) - Roaming Extensions (A user can go to any office and log in to a phone – hopefully check voice mail as well) Basically, I am asking if Asterisk can be a system that will seamlessly operate as one big system and handle failovers as well. After spending hours playing with Asterisk, reading voip-info.org, and watching this list, it seems that Asterisk can handle anything. I just would like re-assurance that I am not chasing a lost cause. A simple Yes or No answer is acceptable to me. Below I have a long version of what I am trying to do if anyone is in the mood to give me more pointers J Brandon (Long Version Follows) Long Story Version: Here is what I have to work with: - Four Offices (One main and three remote) - Dedicated T1 lines connecting three remote offices to one main office (all connections made through the main office) - Will have a T1 Voice line at the main office - Three PSTN lines at each remote office Essentially what I would like to do is create a system comparable to the ShoreTel ShoreGear product line (if you are familiar with it). This system will seamlessly unite all offices as one and provide failover in the case of line outage. It also allows users to roam from phone to phone across offices seamlessly. It has many more features, but those are two main features I am looking for. About 40 total phones will be deployed. To make it even more difficult, I would like all user extensions to start the same (i.e. across offices all extensions are 5### with no discernable pattern). Progress so far: At this time I have determined that I will need a server at each office as well as a T1 card (TE110P) at the Main office and the four port TDM PSTN cards at each remote office. I plan on using the Polycom IP 430 or 501 (Undecided, 501 if required). I have been using TrixBox to this point, would like to continue if possible. It appears that I will want to use DunDi in some fashion to unite the branches. My main roadblock right now is trying to figure out how to get all the information across the offices at the same time (extensions, voicemail). I have successfully had two boxes communicate, but what I am looking for is much more complex I feel. I have thought of synchronized MySQL databases, but I do not know if that will work the way I wish. If anyone reads this far ;) I am looking for suggestions for routes I might consider or places I should/could look for more information. I am relatively new to Asterisk, but I am not afraid to get my hands dirty. If something I said did not make any sense or if there is more information I could provide, I am happy to help where I can. Thank you for your time and assistance. Brandon Comouche An IT Guy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ New Yahoo! Mail is the ultimate force in competitive emailing. Find out
Re: [asterisk-users] Best FXO Gateway
Linksys SPA400 is a 4 port FXO gateway. Cameron ___ New Yahoo! Mail is the ultimate force in competitive emailing. Find out more at the Yahoo! Mail Championships. Plus: play games and win prizes. http://uk.rd.yahoo.com/evt=44106/*http://mail.yahoo.net/uk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playback uses channel's language, background doesn't
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetLanguage
There you can found how you can get the current language ( the same
used by playback ), so you can set a local variable to the current
language and use it instead of the blank value
This works:
exten =
98764,1,Background(to-listen-to-it|m|${LANGUAGE()}|macro-systemrecording)
Wiki updated.
Thanks
Cameron
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Re: [asterisk-users] How to get values of local channels context
Check out /path/to/src/asterisk/doc/README.variables
${DIALEDPEERNUMBER} would give it to me if I sliced it up.
exten = s,n,Set(Foo=${CUT(DIALEDPEERNUMBER,@,2)})
exten = s,n,Set(Foo=${CUT(Foo,/n,1)})
Are there any better options?
Cameron
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Re: [asterisk-users] How to get values of local channels context
Check out /path/to/src/asterisk/doc/README.variables
${DIALEDPEERNUMBER} would give it to me if I sliced it up.
exten = s,n,Set(Foo=${CUT(DIALEDPEERNUMBER,@,2)})
exten = s,n,Set(Foo=${CUT(Foo,/n,1)})
${CHANNEL} gets me something similar.
Too bad I now have to rename my contexts. Putting a - in them makes using Cut
with ${CHANNEL} difficult!
Cameron
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[asterisk-users] Playback uses channel's language, background doesn't
I have the following in the dialplan:
[macro-systemrecording]
exten = s,1,Goto(${ARG1},1)
exten = dorecord,1,Record(/tmp/${CALLERID(number)}-ivrrecording:wav)
exten = dorecord,n,Wait(1)
exten = dorecord,n,Goto(confmenu,1)
exten = docheck,1,Playback(/tmp/${CALLERID(number)}-ivrrecording)
exten = docheck,n,Wait(1)
exten = docheck,n,Goto(confmenu,1)
exten =
confmenu,1,Background(to-listen-to-itpress-1to-rerecord-itpress-star|m||macro-systemrecording)
exten = confmenu,n,Read(RECRESULT||1|||4)
exten = confmenu,n,GotoIf($[x${RECRESULT}=x*]?dorecord,1)
exten = confmenu,n,GotoIf($[x${RECRESULT}=x1]?docheck,1)
exten = confmenu,n,Goto(1)
exten = 1,1,Goto(docheck,1)
exten = *,1,Goto(dorecord,1)
exten = t,1,Playback(goodbye)
exten = t,n,Hangup
exten = i,1,Playback(pm-invalid-option)
exten = i,n,Goto(confmenu,1)
exten = h,1,Hangup
When this is called the following is shown in the CLI
-- Goto (macro-systemrecording,docheck,1)
-- Executing Playback(SIP/223344-0928bbb8, /tmp/2595-ivrrecording) in
new stack
-- Playing '/tmp/2595-ivrrecording' (language 'nz')
-- Executing Wait(SIP/223344-0928bbb8, 1) in new stack
-- Executing Goto(SIP/223344-0928bbb8, confmenu|1) in new stack
-- Goto (macro-systemrecording,confmenu,1)
-- Executing BackGround(SIP/223344-0928bbb8,
to-listen-to-itpress-1to-rerecord-itpress-star|m||macro-systemrecording)
in new stack
-- Playing 'to-listen-to-it' (language '')
As can be seen, Playback uses the channel's language 'nz' but BackGround does
not. Could anyone advise what I'm doing wrong?
Thanks
Cameron
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[asterisk-users] How to get values of local channels context
The variable ${CONTEXT} stores the value of the current context. However if we
are in a macro that will be the name of the macro. How do I access the name of
the local channel's context.
For example:
[macro-test]
exten = s,n,NoOp(Context ${CONTEXT})
CLI shows:
-- Executing NoOp(Local/[EMAIL PROTECTED],2, Context macro-test) in new
stack
I want to get 116-2000 somehow.
Any suggestions would be appreciated.
Cameron
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Re: [asterisk-users] Playback uses channel's language, background doesn't
it may be a bug, try creating a simple test script with only 2 extensions, one with playback the other one with background and see how it works, also post here the asterisk version you are using. Asterisk 1.2.13 exten = 98765,1,Playback(to-listen-to-it) exten = 98764,1,Background(to-listen-to-it|m||macro-systemrecording) exten = 98763,1,Background(to-listen-to-it) -- Executing Playback(SIP/112233-09289b40, to-listen-to-it) in new stack -- Playing 'to-listen-to-it' (language 'nz') -- Executing Hangup(SIP/112233-09289b40, ) in new stack == Spawn extension (116-2000, h, 1) exited non-zero on 'SIP/112233-09289b40' -- Executing BackGround(SIP/112233-09289b40, to-listen-to-it|m||macro-systemrecording) in new stack -- Playing 'to-listen-to-it' (language '') -- Executing Hangup(SIP/112233-09289b40, ) in new stack == Spawn extension (116-2000, h, 1) exited non-zero on 'SIP/112233-09289b40' -- Executing BackGround(SIP/112233-09289b40, to-listen-to-it) in new stack -- Playing 'to-listen-to-it' (language 'nz') -- Executing Hangup(SIP/112233-09289b40, ) in new stack == Spawn extension (116-2000, h, 1) exited non-zero on 'SIP/112233-09289b40' So it seems assume that since I passed a blank language override to the Background application, that I want a blank language. Any ideas on how to get background to use the default language? Regards Cameron ___ What kind of emailer are you? Find out today - get a free analysis of your email personality. Take the quiz at the Yahoo! Mail Championship. http://uk.rd.yahoo.com/evt=44106/*http://mail.yahoo.net/uk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to get values of local channels context
CLI shows:
-- Executing NoOp(Local/[EMAIL PROTECTED],2, Context macro-test) in
new stack
I want to get 116-2000 somehow.
Any suggestions would be appreciated.
So use ${MACRO_CONTEXT} .
Thanks
But doesn't this give the calling context which, if itself is another macro,
will still not give me what I want? If macro-test is called by macro-first then
${MACRO_CONTEXT} = macro-first. Surely there's a way to get the context
directly from the Local channel itself?
Cameron
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[asterisk-users] Destroy a zombie sip channel
I am unable how to get a zomebie sip channel to hangup. I've tried the following in the manager but it doesn't work. Action: Status Response: Success Message: Channel status will follow Event: Status Privilege: Call Channel: SIP/2003-09e2bbe8ZOMBIE CallerID: 093611168 CallerIDName: unknown Account: State: Up Link: SIP/2003-09e719f0 Uniqueid: 1171346560.592 Event: StatusComplete Action: Hangup Channel: SIP/2003-09e2bbe8ZOMBIE Response: Success Message: Channel Hungup Action: Status Response: Success Message: Channel status will follow Event: Status Privilege: Call Channel: SIP/2003-09e2bbe8ZOMBIE CallerID: 093611168 CallerIDName: unknown Account: State: Up Link: SIP/2003-09e719f0 Uniqueid: 1171346560.592 Event: StatusComplete Any other suggestions for how to kill this thing (ideally without restarting asterisk) would be appreciated. Cameron ___ What kind of emailer are you? Find out today - get a free analysis of your email personality. Take the quiz at the Yahoo! Mail Championship. http://uk.rd.yahoo.com/evt=44106/*http://mail.yahoo.net/uk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No RTP packets received by Asterisk when calling SIP to SIP
I have the following setup: UA1 (SPA2000) -- Nat1 -- Asterisk (public internet) -- Nat 1 -- UA2 (X-Lite) Relevant parts of sip.conf [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) externip = 60.234.100.100 ;External IP address localnet = 192.168.1.0/255.255.255.0;Local network address allow=all [1590] username=1590 type=friend secret=secret qualify=no port=5060 nat=yes [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=test canreinvite=no allow=all [1593] username=1593 type=friend secret=secret qualify=no port=5060 nat=yes [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=test canreinvite=no allow=all I have enabled rtp debugging and notice that Asterisk is receiving no rtp traffic. When I call from either UA to voicemail for example I see RTP traffic e.g. call from 1590 Got RTP packet from 60.234.200.200:38510 (type 0, seq 1245, ts 207620, len 160) Sent RTP packet to 60.234.200.200:38510 (type 0, seq 61963, ts 34880, len 160) e.g. call from 1593 Got RTP packet from 60.234.200.200:16470 (type 0, seq 892, ts 316685167, len 240) Sent RTP packet to 60.234.200.200:16470 (type 0, seq 1156, ts 15360, len 160) I thought that with canreinvite=no all audio would go through Asterisk. What have I missed? Asterisk 1.2.13 Fedora Core 5 Regards Cameron ___ Copy addresses and emails from any email account to Yahoo! Mail - quick, easy and free. http://uk.docs.yahoo.com/trueswitch2.html___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE: Realtime Voicemail Password Change Not Working
I was able to update the password through the dialplan with this:
exten = ,1,MYSQL(Connect connid 127.0.0.1 pbx pbx pbxdb)
exten = ,2,MYSQL(Query resultid ${connid} UPDATE\ voicemail\ SET\
password=\ where\ mailbox=52007)
exten = ,3,MYSQL(Clear ${resultid})
exten = ,4,MYSQL(Disconnect ${connid})
exten = ,5,Hangup
Finaly I got an update statement in the mysql log:
12 Query UPDATE voicemail SET password= where mailbox=52007
So these results suggest that mysql, voicemail table, and the res_mysql
adddon are working fine. It suggests that app_voicemail is not passing
the
update statement to the res_mysql driver.
This was a clean install, nothing out of the ordinary.
I would second the other posters suggestion: use Realtime update (show
application realtime update) since it uses the actual realtime setup. The
MySQL command shown above uses a new connection that you specify so is not
such a good test.
Cameron
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Re: [asterisk-users] Zap channels staying offhook - restart required
Anyway, my question is, how do I get the offhook status to reset? So far only a server reboot works. I tried: - physically disconnecting the line from the socket - restarting asterisk - zap destroy channel and restarting asterisk Any suggestions on how to avoid a reboot? I tried the following: unload chan_zap.so load chan_zap.so That seemed to reset the offhook status without a reboot. How do I access this in a dialplan or via the Manager? Thanks Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime sipusers and rtcachefriends... bigheadache!!
- Original Message - From: kjcsb [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, January 24, 2007 8:24 AM Subject: Re: [asterisk-users] realtime sipusers and rtcachefriends... bigheadache!! hi folks, I am using asterisk 1.2.13 (debian etch). My customer's sip accounts are stored in realtime sipusers. I have enabled in sip.conf rtcachefriends=yes and ignoreregexpire=yes Each account has nat=yes Now, I have lot of problems. for example, when I change the 'secret' field of a user in the database, it doesn't get reflected in Asterisk, who is still expecting the old password. As far as I know when rtcachefriends=yes database changes are unavailable to Asterisk until a reload is performed. Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Realtime - one database driver, multiple databases
Is it possible to have different families refer to different databases for the same database driver? The examples I have seen specify the same host, database etc. For example is this possible: extconfig.conf sipusers = mysql,asterisk,asterisk_sip voicemail = mysql,mail,voicemail If it is possible, what is the correct way to specify the details in res_mysql.conf? Something like this? [general] dbhost = asterisk.domain.com dbname = asterisk dbuser = asteriskuser dbpass = test dbport = 3306 dbhost = mail.domain.com dbname = mail dbuser = mailuser dbpass = test dbport = 3306 Regards Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zap channels staying offhook - restart required
I have a situation where the two Zap channels on a TDM400 are staying offhook after a random period of time; it is not (I believe) related to the FXO side not hanging up. Actually I suspect the server is overheating but I need to do more analysis. Anyway, my question is, how do I get the offhook status to reset? So far only a server reboot works. I tried: - physically disconnecting the line from the socket - restarting asterisk - zap destroy channel and restarting asterisk Any suggestions on how to avoid a reboot? Also suggestions on debugging this would be appreciated. Regards Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime sipusers and rtcachefriends... bigheadache!!
hi folks, I am using asterisk 1.2.13 (debian etch). My customer's sip accounts are stored in realtime sipusers. I have enabled in sip.conf rtcachefriends=yes and ignoreregexpire=yes Each account has nat=yes Now, I have lot of problems. for example, when I change the 'secret' field of a user in the database, it doesn't get reflected in Asterisk, who is still expecting the old password. As far as I know when rtcachefriends=yes database changes are unavailable to Asterisk until a reload is performed. Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error on answer a SIP 401 message
I'm a voip service provider and i'm setting up a asterisk box to register around 100 lines from my central softswitch. This asterisk box will be placed inside a customer and has a digium card to be interconected with customer's pabx. My problem is that when asterisk send register message, my softswitch return with sip 401 and asterisk should send a register message with Authorization in header. Only after 10, or 15 or 30 or 43 or 2 messages 401 asterisk start to send Authorization in header. This is a random time, don't follow any rule. I had something vaguely similar. Asterisk was replying on the wrong interface/network card. Might be worth checking. Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Billing solution
Can anyone recommend a call accounting solution with rating for post paid billing that works well with asterisk using the account code or any other info from the CDR? I don't want the billing software to any phone calls for me, therefore any solution that modifies my extensions.conf is out, nor does it have to allow for customers the ability to log in to check their usage/balances. I have looked at astbill but it looks to be way overcomplicated for what I want, as well as it requires realtime. Thank you CDRTool does call rating Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Input on Dundi
The RealTime command pulls all the entire record from the database and prepends all the fields with the last argument (here is have DN_) so when the record is pulled, all the records info is available as a variable like DN_port and DN_ipaddr. This is a really cool command. Hope this helps. Wow, thanks for the examples JR. This is exactly what I needed. I was not aware of the RealTime command. That will be very useful. I also stumbled across RealTimeUpdate recently and documented it on the wiki. Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 5.8gig phone MWI
Does anyone have personal experience with a 5.8gig wireless phone (system) that has an MWI that WORKS with asterisk via fxs (in my case spa3k) generated MWI. I know the spa3k does stuttered dialtone but not sure if it generates FSK MWI. Uniden DSS7815 MWI works with SPA3K. Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Best way to access MySQL data from dial plan
Resending as message didn't show up the first time
I need to access MySQL from the dial plan. Currently I am using the MYSQL
function:
exten = *78,n,MYSQL(Connect asterisklocal localhost asteriskuser password
asterisk)
exten = *78,n,MYSQL(Query resultid ${asterisklocal} CALL\
sp_ins_into_avp(\'/DND/${CALLERID(number)}\'\,\'YES\'))
exten = *78,n,MYSQL(Clear ${resultid})
exten = *78,n,MYSQL(Disconnect ${asterisklocal})
This shows authentication details in the Asterisk CLI which is not ideal.
What is the recommended way to access MySQL data?
Asterisk 1.2
CentOS 4.4
MySQL 5.0
Regards
Cameron
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Re: [asterisk-users] Re: Best way to access MySQL data from dial plan
I'm not sure that any solution with the MySQL dialplan command is going to be ideal. You also can't nest your queries, ie the connectid/result id seems to only be good for one resultset at a time... try doing something like findme/followme with that! Thanks What is a better way to do it then in terms of performance, security, and flexibility? Using exec and a shell script, or agi or something else? Regards Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MWI, Realtime SIP, Voicemail and Extensions, UAs registered with SER
I have the following setup: - UAs registered with SER/OpenSER - SIP peers (non cached), extensions, voicemail setup (not message storage) defined in Asterisk 1.2 using Realtime When a message is left in the user's mailbox, no Notify message is sent to SER. 1. If the SIP peer is defined in sip.conf with a host=ser.domain.com then the notfy is sent to SER. 2. If realtimecache=yes is set in sip.conf and the SIP peer is visible with sip show peers and the SIP peer host field is set to ser.domain.com then the notify is sent to SER. I have read numerous articles regarding this including: - the posting http://www.asteriskguru.com/board/need-help-for-voicemail-notification-vt535.html refers to a patch noted on http://www.voip-info.org/wiki-Asterisk+at+large. The patch is listed under Method 3, which relies on sip peers being defined in sip.conf i.e. it doesn't work for non cached realtime. - Method 1 of http://www.voip-info.org/wiki-Asterisk+at+large describes a way to send the Notify direct to the SIP UA. This relies on the phone contact details (e.g. IP address) being defined in sip.conf - not applicable in my case. - Method 2 of http://www.voip-info.org/wiki-Asterisk+at+large relates to SIP UAs registered with SER and states that Asterisk sends NOTIFY only to UACs that are registered at the Asterisk. This is not the case as described in 1 above and Method 5 of Asterisk-at-large. - Method 4 of http://www.voip-info.org/wiki-Asterisk+at+large assumes cached SIP realtime peers. I don't want to cache. - the posting http://forums.digium.com/viewtopic.php?t=4363highlight relates to SIP UAs registered with Asterisk, not those registered with SER. - the article http://openser.org/dokuwiki/doku.php/asterisk:realtime-integration doesn't deal with MWI. - the posting http://asterisk.mdaniel.net/?p=14 creates the msgnum.txt file on a remote Asterisk server and so is not relevant to my scenario. Can anyone advise how they are sending SIP Notify messages from Asterisk to SER for non-cached realtime SIP peers? Regards Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MWI, Realtime SIP, Voicemail and Extensions, UAs registered with SER
I have the following setup: - UAs registered with SER/OpenSER - SIP peers (non cached), extensions, voicemail setup (not message storage) defined in Asterisk 1.2 using Realtime When a message is left in the user's mailbox, no Notify message is sent to SER. 1. If the SIP peer is defined in sip.conf with a host=ser.domain.com then the notfy is sent to SER. 2. If realtimecache=yes is set in sip.conf and the SIP peer is visible with sip show peers and the SIP peer host field is set to ser.domain.com then the notify is sent to SER. I have read numerous articles regarding this including: - the posting http://www.asteriskguru.com/board/need-help-for-voicemail-notification-vt535.html refers to a patch noted on http://www.voip-info.org/wiki-Asterisk+at+large. The patch is listed under Method 3, which relies on sip peers being defined in sip.conf i.e. it doesn't work for non cached realtime. - Method 1 of http://www.voip-info.org/wiki-Asterisk+at+large describes a way to send the Notify direct to the SIP UA. This relies on the phone contact details (e.g. IP address) being defined in sip.conf - not applicable in my case. - Method 2 of http://www.voip-info.org/wiki-Asterisk+at+large relates to SIP UAs registered with SER and states that Asterisk sends NOTIFY only to UACs that are registered at the Asterisk. This is not the case as described in 1 above and Method 5 of Asterisk-at-large. - Method 4 of http://www.voip-info.org/wiki-Asterisk+at+large assumes cached SIP realtime peers. I don't want to cache. - the posting http://forums.digium.com/viewtopic.php?t=4363highlight relates to SIP UAs registered with Asterisk, not those registered with SER. - the article http://openser.org/dokuwiki/doku.php/asterisk:realtime-integration doesn't deal with MWI. - the posting http://asterisk.mdaniel.net/?p=14 creates the msgnum.txt file on a remote Asterisk server and so is not relevant to my scenario. Can anyone advise how they are sending SIP Notify messages from Asterisk to SER for non-cached realtime SIP peers? Regards Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE: Best way to access MySQL data from dial plan
Setup extconfig to have realtime access to the database/table you want to pull info from, then in the dialplan use the app Realtime. Thanks. I didn't know that you could use RealTime in the dialplan like that. I thought is was just for sip, extensions etc. I created a wiki page at http://www.voip-info.org/wiki/view/Asterisk+cmd+RealTime. Feel free to edit if it's wrong! Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Resolved: Re: [asterisk-users] Modprobe zaptel reports FATAL: Module zaptel notfound
There was something screwy going on with kernel vs kernel-devel. So I rolled back to kernel-*-2.6.9-42 rather than kernel-*-2.6.9-42.0.3. Zaptel has now installed successfully. I don't believe this is a problem with 2.6.9-42.0.3 per se. Rather my system had different versions of kernel vs kernel-devel. Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Modprobe zaptel reports FATAL: Module zaptel not found
I am (unsuccessfully) trying to install zaptel (incl ztdummy - I don't have any Digium hardware) on CentOS 4. uname -r 2.6.9-42.ELsmp Not sure how this relates to 2.6.9-42.0.3 (see below) ln -s /usr/src/kernels/`uname -r` /usr/src/linux ln -s /usr/src/kernels/`uname -r` /usr/src/linux-2.6 cd /usr/src/zaptel-1.2.11* make linux26 You do not appear to have the sources for the 2.6.9-42.ELsmp kernel installed. rpm -q kernel-devel kernel-devel-2.6.9-42.EL kernel-devel-2.6.9-42.0.3.EL I don't understand why there are two kernel-devel packages installed rpm -q kernel-smp-devel kernel-smp-devel-2.6.9-42.0.3.EL rm /usr/src/linux rm /usr/src/linux-2.6 ln -s /usr/src/kernels/2.6.9-42.0.3.EL-smp-i686 /usr/src/linux ln -s /usr/src/kernels/2.6.9-42.0.3.EL-smp-i686 /usr/src/linux-2.6 cd /usr/src/zaptel-1.2.11* make linux26 make install make config /sbin/modinfo zaptel modinfo: could not find module zaptel find /lib/modules | grep zaptel /lib/modules/2.6.9-42.0.3.ELsmp/extra/zaptel.ko cd /usr/src/kernels ls 2.6.9-42.0.3.EL-hugemem-i686 2.6.9-42.0.3.EL-i686 2.6.9-42.0.3.EL-smp-i686 2.6.9-42.EL-i686 Could anyone shed any light on what I've done wrong? Thanks Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Modprobe zaptel reports FATAL: Module zaptel notfound
ln -s /usr/src/kernels/`uname -r` /usr/src/linux ln -s /usr/src/kernels/`uname -r` /usr/src/linux-2.6 Unnecessary. Just install the relevant kernel-devel package. What instructions are you following? I already had kernel-devel installed and was still getting the message You do not appear to have the sources for the 2.6.9-42.ELsmp kernel installed. So I hunted around and found a link indicating that *possibly* I needed to create a symbolic link. I tried this in two ways: ln -s /usr/src/kernels/`uname -r` /usr/src/linux etc Got the same message at make linux26 ln -s /usr/src/kernels/2.6.9-42.0.3.EL-smp-i686 /usr/src/linux etc make linux26 was successful but modprobe can't find zaptel What is the output of: uname -r 2.6.9-42.ELsmp If you run 'depmod', does it change anything? No Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Desktop integration
http://activa.sourceforge.net/ does this. - Original Message - From: Ondrej Valousek To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, November 14, 2006 12:28 AM Subject: [asterisk-users] Desktop integration Hi all, I am interested in integrating my telephone system (I am using hardphones and Asterisk) with my desktop - something like this: 1. someone sends me his/her phone number via email/icq 2. I cut/paste the number in some application/web page (?) 3. my phone starts ringing and when I pick it up I will get connected with the remote party. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk to listen for sip traffic on 80 and 5060
I have Asterisk listening for sip traffic on port 5060. I want to allow users to use either port 80 or 5060 if they want. Hopefully this will avoid some firewall issues. Is this a sensible/crazy thing to do? I have done a bunch of searching and believe iptables can help but haven't been able to find an example to forward something from 80 to 5060 inbound and outbound where iptables is running on the same machine as Asterisk. Is iptables the best way to do it (without other hardware) or is there an alternative? If anyone has used iptables to do this would you be willing to share the setup? Would something like ths work for inbound?: iptables -t nat -A PREROUTING -p udp --dport 80 --sport 1024:65535 -j DNAT --to 127.0.0.1:5060 iptables -A FORWARD -p udp -d 1270.0.1 \ --dport 5060 -m state --state NEW -j ACCEPT iptables -A FORWARD -t filter -m state \ --state NEW,ESTABLISHED,RELATED -j ACCEPT What about outbound? Alternatively is there a better option? Any suggestions appreciated. Regards Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] several behind NAT
Also, where can I get information on provisioning? These phones will be out of my hands soon and I'd like to be able to update the configs. I saw a few utilities for generating the configs, but I'd like more specific info - I don't mind editing files by hand but want to know how it works. Does anyone have some resources? Check the Grandstream website for a java-based provisioning tool for Linux Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dial D option with w for wait?
- Original Message - From: BerkHolz, Steven [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, November 01, 2006 1:27 AM Subject: [asterisk-users] dial D option with w for wait? From WIKI: D(digits): After the called party answers, send digits as a DTMF stream, then connect the call to the originating channel. (You can also use 'w' to produce .5 second pauses.) When I use the D option to send a call to my paging system and pick a zone, the Tone is too early. I have tried the 'w' option, but it does not appear to work. No matter how many 'w's I use, the tone is still immediately on answer. Did you find a resolution to this? Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] light web user interface
FreePBX allows you to specify an extension range per login so that only extensions within the range are visible to that user. Cameron - Original Message - From: Curt Shaffer To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Wednesday, November 01, 2006 1:40 AM Subject: RE: [asterisk-users] light web user interface Basically I would like a page that would allow a user to log in and modify their extension only. So for example, I log in for extension 102 once in there I can turn on or off my call waiting. Add a number to call forward to. Change the email address my voice mail gets sent to. Add any numbers I may want to block via caller ID. Maybe view my voice mails that are saved and be able to download them in wav format from there. Add find me follow me extensions and numbers, etc I would also like it open enough that I can add features to it. Im not the best at PHP but I can work my way around in it. I thought maybe freePBX allowed this with its users but I cant see where you can lock them down to only see information on a particular extension. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid BSent: Tuesday, October 31, 2006 3:44 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] light web user interface What attributes are you talking about ? Depending on what they are it may be real simple to set something up. - Original Message - From: Curt Shaffer To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Monday, October 30, 2006 9:51 PM Subject: [asterisk-users] light web user interface Does anyone know of a really lightweight web interface that allows users to log in and modify attributes of their extension only? Thanks Curt ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone got a dialplan for SPA ATAs for ISN?
Googling for a while has turned up evidence that this can be corrected by a carefully-crafted dialplan for the Sipuras, at least, but the avaialable documentation is, let's say, a little convoluted. Try this on Sipura (*x.*x.) Seemed to work for me. Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Increase VoiceMail Messages Recording Gain - AudioCalls are Ok
The problem is:Right now, and i'm referring only to calls directly handled by VoiceMail application, the users get their audio files in email but the audio is very very low. I've thought about changing RX gain on PRI interface between legacy pbx and asterisk, but until now no complaining with audio calls. There's a patch for this: http://bugs.digium.com/file_download.php?file_id=10824type=bug Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Segmentation fault on Asteriskstartup:res_config_mysql.so problem?
Did you do a make make install for add-ons BEFORE doing
so for asterisk?
If so try asterisk first and when all is installed install
add-ons.
--
I tried a make clean make make install for asterisk and then for
asterisk-addons but am still getting the segmentation fault on asterisk
startup. rm res_config_mysql.so allows Asterisk to start.
Still trying...
mkdir /usr/lib/asterisk.backup.20060928
mv /usr/lib/asterisk/* /usr/lib/asterisk.backup.20060928
mkdir /usr/include/asterisk.backup.20060928
mv /usr/include/asterisk/* /usr/include/asterisk.backup.20060928/
cd /usr/src/asterisk-1.2.12.1
make clean make make install
cd /usr/src/asterisk-addons-1.2.4
perl -p -i.bak -e
's/CFLAGS.*D_GNU_SOURCE/CFLAGS+=-D_GNU_SOURCE\nCFLAGS+=-DMYSQL_LOGUNIQUEID/'
Makefile
make clean make make install
Install logs look fine.
STARTING ASTERISK
/usr/sbin/safe_asterisk: line 40: 6631 Segmentation fault (core
dumped) asterisk ${CLIARGS} ${ASTARGS} /dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
/usr/sbin/safe_asterisk: line 40: 6690 Segmentation fault (core
dumped) asterisk ${CLIARGS} ${ASTARGS} /dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
rm res_config_mysql.so allows Asterisk to start.
Any advice appreciated.
Cameron
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Re: [asterisk-users] Segmentation fault on Asterisk startup:res_config_mysql.so problem?
Did you do a make make install for add-ons BEFORE doing so for asterisk? If so try asterisk first and when all is installed install add-ons. -- I tried a make clean make make install for asterisk and then for asterisk-addons but am still getting the segmentation fault on asterisk startup. rm res_config_mysql.so allows Asterisk to start. Any other suggestions appreciated. Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Segmentation fault on Asterisk startup: res_config_mysql.so problem?
When Asterisk starts I get a Segmentation fault
/usr/sbin/safe_asterisk: line 40: 30548 Segmentation fault (core
dumped) asterisk ${CLIARGS} ${ASTARGS} /dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
If I remove /usr/lib/asterisk/modules/res_config_mysql.so Asterisk starts
normally.
tail /var/log/asterisk/full.log
Sep 24 15:46:05 VERBOSE[30608] logger.c: == Parsing
'/etc/asterisk/res_mysql.conf': Sep 24 15:46:05 VERBOSE[30608] logger.c:
== Parsing '/etc/asterisk/res_mysql.conf': Found
Sep 24 15:46:05 WARNING[30608] res_config_mysql.c: MySQL RealTime: No
database socket found, using '/tmp/mysql.sock' as default.
Sep 24 15:46:05 DEBUG[30608] res_config_mysql.c: MySQL RealTime Host:
127.0.0.1
Sep 24 15:46:05 DEBUG[30608] res_config_mysql.c: MySQL RealTime Port: 3306
Sep 24 15:46:05 DEBUG[30608] res_config_mysql.c: MySQL RealTime User: root
Sep 24 15:46:05 DEBUG[30608] res_config_mysql.c: MySQL RealTime Password:
password
vi /etc/asterisk/res_mysql.conf
[general]
dbhost = 127.0.0.1
dbname = asterisk
dbuser = root
dbpass = password
dbport = 3306
;dbsock = /var/lib/mysql/mysql.sock
If I uncomment the dbsock line I get the same result (although the database
socket warning is not displayed in the log file).
I am using MySQL for CDR logging so I don't think it's a MySQL problem.
Asterisk 1.2.12.1
Asterisk addon 1.2.4
When I install Asterisk I receive a warning:
Your Asterisk modules directory, located at /usr/lib/asterisk/modules
contains modules that were not installed by this version of Asterisk.
However I cleared out the /usr/lib/asterisk/modules directory before make
clean make make install for both add-ons and asterisk so I'm a bit
mystified by that.
Could anyone suggest further checks I could do?
Thanks
Cameron
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[asterisk-users] How does Asterisk determine an incoming SIP Channel name?
I have a number of different calls coming in to Asterisk from one SIP proxy. All calls are currently allocated the same SIP Channel name but I want them to be named differently. Note that Asterisk registers with the SIP Proxy, not the other way around. sip.conf register=5551234:[EMAIL PROTECTED]/5551234 register=5552345:[EMAIL PROTECTED]/5552345 register=5553456:[EMAIL PROTECTED]/5553456 register=5554567:[EMAIL PROTECTED]/5554567 [5551234 (Accounts)] username=5551234 type=peer host=domain.com [5553456 (Sales)] username=5553456 type=peer host=domain.com [5554567 (Support)] username=5554567 type=friend host=domain.com [5552345] username=5552345 type=friend host=domain.com When a call comes in from the host to 5551234 for example, the channel is named SIP/5552345-b7b0b8a8. The same name is given a call from 5553456. If I remove the [5552345] entry from sip.conf then the channel is named SIP/5554567-b7b0b8a8 i.e. the channel is named according to the first username for the host starting at the bottom of the sip.conf file. Could anyone suggest how I can achieve the desired result? Thanks and regards Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Register message received from realtime peer crashes Asterisk
When Asterisk (1.2.12.1) receives a SIP register message for a realtime peer, the CLI reports Disconnected from Asterisk server. Asterisk has disappeared: asterisk -r Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) A look at the full log doesn't reveal much: Sep 17 06:11:25 DEBUG[11011] acl.c: # Testing 60.234.nnn.nnn with 192.168.1.0 Sep 17 06:11:25 DEBUG[11011] chan_sip.c: Target address 60.234.nnn.nnn is not local, substituting externip Sep 17 06:11:25 DEBUG[11011] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '6000' Sep 17 06:11:25 DEBUG[11011] res_config_mysql.c: MySQL RealTime: Everything is fine. Asterisk then restarts (it gets a new pid) and will continue running happily until a new register request is received for a realtime peer. Note that Asterisk operates normally in all other respects until the register is received e.g. sip peers in sip.conf can register and make calls successfully. Only when a register is received from a peer that exists in sip_buddies does Asterisk crash. I can run the query successfully on mysql command line: SELECT * FROM sip_buddies WHERE name = '6000'; snip 1 row in set (0.32 sec) A review of syslog and the mysql log reveals little: mysql log 060917 6:54:31 14 Init DB asterisk 14 Query SELECT * FROM sip_buddies WHERE name = '6000' syslog Nothing report at the time of the crash (06:54). extconfig.conf [settings] sipusers = mysql,asterisk,sip_buddies sippeers = mysql,asterisk,sip_buddies res_mysql.conf [general] dbhost = 127.0.0.1 dbname = asterisk dbuser = root dbpass = password dbport = 3306 Could anyone advise what's going on or further checking that I could do to analyse the problem? Thanks Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP_HEADER function; what names are available?
I have read the wiki about the SIP_HEADER function (http://www.voip-
info.org/wiki/index.php?page=Asterisk+func+sip_header). Where can I get
a list of the names that are available to be used with the function e.g.
TO is one name as in ${SIP_HEADER(TO)}. What are the others?
I would guess that you can check the RFC. Easier is to turn on SIP debug
and see the INVITE packet yourself and
check the headers that you have with your equipment.
/Olle
Thanks but I don't know how to get the actual INVITE details (the request
URI?). For example I want to get sip:[EMAIL PROTECTED] SIP/2.0 from
the following dialogue:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Record-Route: sip:147.202.nn.nnn;ftag=bf7eced18eb7271b;lr=on
Via: SIP/2.0/UDP 147.202.nn.nnn;branch=z9hG4bKe49c.21b320a3.0
Via: SIP/2.0/UDP 60.234.nnn.nnn;branch=z9hG4bK76bf3dec8d45b972
From: User sip:[EMAIL PROTECTED];tag=bf7eced18eb7271b
To: sip:[EMAIL PROTECTED]
etc
I can get Record-Route, Via, From, To etc but don't know how to get the bit
after the INVITE. Interestingly only the first Via is returned by
${SIP_HEADER(VIA)}.
I've tried R-URI, RURI, URI, ALL, *, blank.
Any advice appreciated.
Cameron
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Re: [asterisk-users] Asterisk Real Time and sip.conf file used at thesame time
I guess my problem might be that, because I pretend Asterisk to use my sip.conf static configuration file and also MySQL tables referenced in extconfig.conf like this: [settings] sipusers = mysql,asterisk,sip sippeers = mysql,asterisk,sip voicemail = mysql,asterisk,voicemail While I'm using one thing I can't use the other right??? Based on my limited knowledge you *can* use both at the same time. Any Sip details created in the sip table in your asterisk database will be available immediately to Asterisk. They will not be reported in the command line if you enter sip show peers. This is sometimes called realtime dynamic. Any Sip entries in sip.conf will *also* be available to Asterisk but only on reload e.g. sip reload. These will be reported in the command line if you enter sip show peers. However to complicate matters further there are two additional things to be aware of: realtime caching realtime static Realtime caching loads the sip details from the database in a similar way to how the details from sip.conf are loaded i.e. both the details from sip.conf and from the database will be reported if you enter sip show peers. However changes made in the database are not immediately available - you need to reload just like if you made a change in sip.conf. To enabled this you must set rtcachefriends=yes in sip.conf Realtime static is totally different to the realtime discussed above. It uses a different database structure and is intended to replace the Asterisk static files. Beyond that I'm unsure. Personally I think realtime is a very misleading name. Extconfig would be a better term. Extconfig allows Asterisk to read its configuration files from any external source. Asterisk can be configured to source certain configuration files (e.g. sip.conf) internally (the default which will read from a text file) or (these are mutually exclusive) from an external source such as a database (so-called realtime static). *In addition*, Extconfig can read configuration information from an external source on-the-fly (realtime dynamic) or cached (realtime cached). If anything I've said above is incorrect I'd sure appreciate an expert correcting me. Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial out based on SIP invite
Assume that I receive an Invite from a SIP device that Asterisk has
registered with. How do I get Asterisk to dial out using the Invite details
as if the Invite had been received from a UA registered with Asterisk? i.e.
UA - SIP Proxy - Asterisk - PSTN gateway.
For example
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Record-Route: sip:147.202.nn.nnn;ftag=bf7eced18eb7271b;lr=on
Via: SIP/2.0/UDP 147.202.nn.nnn;branch=z9hG4bKe49c.21b320a3.0
Via: SIP/2.0/UDP 60.234.nnn.nnn;branch=z9hG4bK76bf3dec8d45b972
From: User sip:[EMAIL PROTECTED];tag=bf7eced18eb7271b
To: sip:[EMAIL PROTECTED]
etc
If the Invite was received from a SIP device registered with Asterisk (in
the [from-internal] context) then the call would be routed to
[outrt-003-test] and dial out correctly.
I want to do the same thing with the Invite received from the SIP proxy. Can
anyone advise how I can achieve this (in Asterisk 1.2.9)? Cut-down versions
of conf files are below.
sip.conf
register=1122334455:[EMAIL PROTECTED]/66554433
[1122334455]
type=peer
host=proxy.domain.com
fromuser=1122334455
context=from-internal
extensions.conf
[from-internal]
include = from-internal-additional
exten = s,1,Macro(hangupcall)
exten = h,1,Macro(hangupcall)
exten = 66554433, 1, ?
[from-internal-additional]
include = outbound-allroutes
[outbound-allroutes]
include = outrt-003-test
exten = foo,1,Noop(bar)
[outrt-003-test]
exten = _90[2-79]XX.,1,Macro(dialout-trunk,1,${EXTEN:1},,)
exten = _90[2-79]XX.,n,Macro(dialout-trunk,5,${EXTEN:1},,)
exten = _90[2-79]XX.,n,Macro(dialout-trunk,3,${EXTEN:1},,)
exten = _90[2-79]XX.,n,Macro(dialout-trunk,2,${EXTEN:1},,)
exten = _90[2-79]XX.,n,Macro(outisbusy,)
[macro-dialout-trunk]
exten = s,1,GotoIf($[${ARG3} = ]?3:2) ; arg3 is pattern password
exten = s,2,Authenticate(${ARG3})
exten = s,3,Macro(user-callerid)
exten = s,4,Macro(record-enable,${CALLERID(number)},OUT)
exten = s,5,Macro(outbound-callerid,${ARG1})
exten = s,6,Set(GROUP()=OUT_${ARG1})
exten = s,7,GotoIf($[ ${GROUP_COUNT()} ${OUTMAXCHANS_${ARG1}} ]?108)
; if we've used up the max channels, continue at (n+101)
exten = s,8,Set(DIAL_NUMBER=${ARG2})
exten = s,9,Set(DIAL_TRUNK=${ARG1})
exten = s,10,AGI(fixlocalprefix) ; this sets DIAL_NUMBER to the proper dial
string for this trunk
exten = s,11,Set(OUTNUM=${OUTPREFIX_${ARG1}}${DIAL_NUMBER}) ; OUTNUM is
the final dial number
exten = s,12,Set(custom=${CUT(OUT_${ARG1},:,1)}) ; Custom trunks are
prefixed with AMP:
exten = s,13,GotoIf($[${custom} = AMP]?16)
exten = s,14,Dial(${OUT_${ARG1}}/${OUTNUM},120,${TRUNK_OPTIONS}) ; Regular
Trunk Dial
exten = s,15,Goto(s-${DIALSTATUS},1)
; This is a custom trunk. Substitute $OUTNUM$ with the actual number and
rebuild the dialstring
; example trunks: AMP:CAPI/:b$OUTNUM$,30,r,
AMP:OH323/[EMAIL PROTECTED]:
exten = s,16,Set(pre_num=${CUT(OUT_${ARG1},$,1)})
exten = s,17,Set(the_num=${CUT(OUT_${ARG1},$,2)}) ; this is where we
expect to find string OUTNUM
exten = s,18,Set(post_num=${CUT(OUT_${ARG1},$,3)})
exten = s,19,GotoIf($[${the_num} = OUTNUM]?20:21) ; if we didn't find
OUTNUM, then skip to Dial
exten = s,20,Set(the_num=${OUTNUM}) ; replace OUTNUM with the actual
number to dial
exten = s,21,Dial(${pre_num:4}${the_num}${post_num},120,${TRUNK_OPTIONS})
exten = s,22,Goto(s-${DIALSTATUS},1)
exten = s,108,Noop(max channels used up)
exten = s-BUSY,1,NoOp(Trunk is reporting BUSY)
exten = s-BUSY,2,Busy()
exten = s-BUSY,3,Wait(60)
exten = s-BUSY,4,NoOp()
exten = _s-.,1,NoOp(Dial failed due to ${DIALSTATUS})
Please note that Asterisk also receives Invites from the same proxy (same IP
and port) that need to be treated differently i.e. as if they were external
incoming calls. If this were not the case then the following sip.conf
achieves the desired result (I've tested this successfully). The call gets
into the from-internal context and the outbound call to the PSTN is made:
sip.conf
register=1122334455:[EMAIL PROTECTED]
[1122334455]
type=peer
context=from-internal
However when I create another SIP peer, even though the Invite from the
Proxy has different From details, and I specify fromuser and host in
sip.conf under [1122334455], the call is treated as an external call.
Any advice appreciated.
Cameron
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[asterisk-users] Accessing SIP URI (not ${SIPURI})
How to I access the URI from an Invite:
INVITE sip:[EMAIL PROTECTED]
I want to set a variable to equal 5556678. The variable ${SIPURI} returns
the From URI.
Regards
Cameron
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[asterisk-users] SIP_HEADER function; what names are available?
I have read the wiki about the SIP_HEADER function
(http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header).
Where can I get a list of the names that are available to be used with the
function e.g. TO is one name as in ${SIP_HEADER(TO)}. What are the others?
Regards
Cameron
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Re: RE : [Asterisk-Users] CDRTool
I have 4.5.4 CDRTool version. I patched my cdr_addon_mysql like this: cd ../asterisk-addons - Add a line into asterisk-addons/Makefile reading: CFLAGS+=-DMYSQL_LOGUNIQUEID - edit cdr_addon_mysql.c and replace the line reading AST_MUTEX_DEFINE_STATIC(mysql_lock); with static ast_mutex_t mysql_lock = AST_MUTEX_INITIALIZER; - change the asterisk table name from cdr to asterisk_cdr in cdr_addon_mysql.c chmod 644 cdr_addon_mysql.so cp cdr_addon_mysql.so /usr/lib/asterisk/modules/ restart Asterisk But when I make , I've got error like this: cdr_addon_mysql.c:61: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) make: *** [cdr_addon_mysql.o] Error 1 rm app_saycountpl.o I had a similar problem and so ignored that patching suggestion. In my testing so far it doesn't seem to have caused a problem. You could post to the cdrtool-users list at freelist.org Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OPENSER / SER and Asterisk
Absolutely. The SER/OpenSER documentation is terrible, and if you post to the OpenSER mailing list, you get very cryptic replies. ___ Whilst I would agree with you regarding SER, the documentation of OpenSER is far better. Documentation of Asterisk Realtime on the other hand. Now *that's* terrible. Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Force immediate re-registration on sip reload
Is there any way to force Asterisk to re-register after a sip reload is issued? At the moment, after a sip reload is issued, sip show registry reports all sip UA entries as Unregistered. How can I get Asterisk to immediately send out a registration request to the proxy? Similarly all SIP peers lose their registration status with Asterisk. So when the device is used to make a call immediately after the SIP reload the call is not processed by Asterisk. It takes about 2 minutes 20 seconds before Asterisk starts processing SIP register requests from UAs and before it sends out the registration request to the proxy. How can I reduce this time? Any suggestions greatfully received. Regards Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA-942 TFTP Provisioning
I'm trying to provision some spa-942 phones via TFTP. The phones get their address from a dhcp server which sends it option 66 (address of the tftp server). After spending some time with the phones and even breaking down to sniff traffic from the phones I see that they are not requesting their config from tftp. I can kind of fake the phones into grabbing their configs by doing something like: A little off-topic but how do you create the provisioning file for Linksys/Sipura devices? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Handling inbound and outbound calls passed from a proxy
I need to handle the following scenarios: 1. UA1 -- SIP Proxy -- Asterisk 2. UA2 -- SIP Proxy -- Asterisk -- PSTN gateway (SIP) I have configured a trunk to register with the SIP proxy: trunk1 register=user1:[EMAIL PROTECTED]/DID1 UA1 calls [EMAIL PROTECTED] and the call is recognised as being to DID1. I set up an inbound route for DID1 and route the call as appropriate. That deals with scenario 1. I then tried to configure another trunk to handle scenario 2: trunk2 context=from-internal host=SIP.Proxy type=peer register=user2:[EMAIL PROTECTED] A call to PSTN1 from the UA is passed to the SIP proxy which recognises it as PSTN call. The SIP proxy updates the From details and passes the call to Asterisk which (I presume) puts the call into the from-internal context and dials the outbound route appropriately. However that setup messes up scenario 1 which now gives a 404 back to UA1. I presume Asterisk is not differentiating between a call made to user1 from UA1 and a call made to PSTN1 from user2. It's just seeing a call from SIP.Proxy and putting it into the from-internal context. Could anyone advise how I would set up Asterisk to cope with both these scenarios? I could setup DID2 but I don't know how to pass the call onto the PSTN gateway. I am using AMP/FreePBX but if someone could advise the general principles I would appreciate it. Thanks Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Determining what gets written to the dst field for a CDR
I have Asterisk set up to write call detail records to MySQL. The number written to the dst field is the number dialled by the user including any prefix (e.g. 12125554433 where 1 gives an outside line). However this is not the number dialled by Asterisk (e.g. in this case Asterisk would drop the 1 and dial 2125554433). Is it possible to write the CDR record with the number dialled by Asterisk rather than that dialled by the user? Any advice appreciated. Regards Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail volume patch
There is a patch available for the quiet voicemail volume issue (bug 6237)
but it isn't intended to work with 1.2.9. The patch below will give you this
functionality for 1.2.9. Add the volgain= parameter to voicemail.conf and
make sure sox is installed.
--- apps/app_voicemail.c.backup 2006-07-18 08:52:14.0 +1200
+++ apps/app_voicemail.c 2006-07-21 08:18:42.0 +1200
@@ -229,6 +229,7 @@
unsigned int flags; /*! VM_ flags */
int saydurationm;
int maxmsg; /*! Maximum number of msgs per folder for this mailbox */
+ double volgain; /*! Volume gain for voicemails sent via email */
struct ast_vm_user *next;
};
@@ -386,6 +387,7 @@
static char externnotify[160];
static char vmfmts[80];
+static double volgain;
static int vmminmessage;
static int vmmaxmessage;
static int maxgreet;
@@ -434,6 +436,7 @@
ast_copy_string(vmu-exit, exitcontext, sizeof(vmu-exit));
if (maxmsg)
vmu-maxmsg = maxmsg;
+ vmu-volgain = volgain;
}
static void apply_option(struct ast_vm_user *vmu, const char *var, const
char *value)
@@ -486,6 +489,8 @@
ast_log(LOG_WARNING, Maximum number of messages per folder is %i.
Cannot accept value maxmsg=%s\n, MAXMSGLIMIT, value);
vmu-maxmsg = MAXMSGLIMIT;
}
+ } else if (!strcasecmp(var, volgain)) {
+ sscanf(value, %lf, vmu-volgain);
} else if (!strcasecmp(var, options)) {
apply_options(vmu, value);
}
@@ -1649,6 +1654,7 @@
char dur[256];
char tmp[80] = /tmp/astmail-XX;
char tmp2[256];
+ char tmpcmd[256];
time_t t;
struct tm tm;
struct vm_zone *the_zone = NULL;
@@ -1774,10 +1780,21 @@
}
if (attach_user_voicemail) {
/* Eww. We want formats to tell us their own MIME type */
- char *ctype = audio/x-;
- if (!strcasecmp(format, ogg))
-ctype = application/;
-
+ char *ctype = (!strcasecmp(format, ogg)) ? application/ :
audio/x-;
+
+ char tmpdir[256], newtmp[256];
+
+ create_dirpath(tmpdir, sizeof(tmpdir), vmu-context, vmu-mailbox,
tmp);
+ snprintf(newtmp, sizeof(newtmp), %s/XX, tmpdir);
+ mkstemp(newtmp);
+ ast_log(LOG_DEBUG, newtmp: %s\n, newtmp);
+ if (vmu-volgain -.001 || vmu-volgain .001) {
+snprintf(tmpcmd, sizeof(tmpcmd), sox -v %.4f %s.%s %s.%s,
vmu-volgain, attach, format, newtmp, format);
+ast_safe_system(tmpcmd);
+attach = newtmp;
+ast_log(LOG_DEBUG, VOLGAIN: Stored at: %s.%s - Level: %.4f - Mailbox:
%s\n, attach, format, vmu-volgain, mailbox);
+ }
+
fprintf(p, --%s\n, bound);
fprintf(p, Content-Type: %s%s; name=\msg%04d.%s\\n, ctype, format,
msgnum, format);
fprintf(p, Content-Transfer-Encoding: base64\n);
@@ -1787,6 +1804,7 @@
snprintf(fname, sizeof(fname), %s.%s, attach, format);
base_encode(fname, p);
fprintf(p, \n\n--%s--\n.\n, bound);
+ unlink(newtmp);
}
fclose(p);
snprintf(tmp2, sizeof(tmp2), ( %s %s ; rm -f %s ) , mailcmd, tmp,
tmp);
@@ -5883,6 +5901,7 @@
char *exitcxt = NULL;
char *extpc;
char *emaildateformatstr;
+ char *volgainstr;
int x;
int tmpadsi[4];
@@ -5919,6 +5938,10 @@
astsearch = no;
ast_set2_flag((globalflags), ast_true(astsearch), VM_SEARCH);
+ volgain = 0.0;
+ if ((volgainstr = ast_variable_retrieve(cfg, general, volgain)))
+ sscanf(volgainstr, %lf, volgain);
+
#ifdef USE_ODBC_STORAGE
strcpy(odbc_database, asterisk);
if ((thresholdstr = ast_variable_retrieve(cfg, general,
odbcstorage))) {
___
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[asterisk-users] 2 NICs; Asterisk receives on eth1 and replies on eth0
I have an Asterisk server with 2 network cards. One provides the LAN connection and the other provides the Internet connection. Currently this is set up in the following way: eth0 192.168.1.5. This provides LAN connectivity eth1 192.168.1.251, gw 192.168.1.252 (Note that other nodes on the network use a different gateway, not 192.168.1.252). This provides the internet connection. The router is set up with DMZ enabled and pointing to 192.168.1.251. Currently, when a SIP device attempts to register across the internet, the following sip dialogue is observed: U 60.234.nnn.nnn:5060 - 192.168.1.251:5060 REGISTER sip:60.234.xxx.xxx SIP/2.0..Via: SIP/2.0/UDP 60.234.nnn.nnn;branch=z9hG4bK9d312477..From: CallerID sip:[EMAIL PROTECTED];tag=bec2273b..To: sip:[EMAIL PROTECTED]..Contact: sip:[EMAIL PROTECTED]..Call-ID: [EMAIL PROTECTED]: 100 REGISTER..Expires: 60..User-Agent: Grandstream HT488 1.0.2.16..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER, OPTIONS,INFO,SUBSCRIBE..Content-Length: 0 # U 192.168.1.5:5060 - 60.234.nnn.nnn:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 60.234.nnn.nnn;branch=z9hG4bK9d312477;received=60.234.nnn.nnn..From: CallerID sip:[EMAIL PROTECTED];tag=bec2273b..To: sip:[EMAIL PROTECTED]..Call-ID: [EMAIL PROTECTED]: 100 REGISTER..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Contact: sip:[EMAIL PROTECTED]..Content-Length: 0 ### U 192.168.1.5:5060 - 60.234.nnn.nnn:5060 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 60.234.nnn.nnn;branch=z9hG4bK9d312477;received=60.234.nnn.nnn..From: CallerID sip:[EMAIL PROTECTED];tag=bec2273b..To: sip:[EMAIL PROTECTED];tag=as23747970..Call-ID: [EMAIL PROTECTED] : 100 REGISTER..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Contact: sip:[EMAIL PROTECTED]..WWW-Authenticate: Digest realm=asterisk, nonce=1442f9e8..Content-Length: 0 where: 60.234.nnn.nnn is public IP of SIP client 60.234.xxx.xxx is public IP of Asterisk server Asterisk seems to be replying on eth0 whereas the inbound traffic was received on eth1. This leads me to think that there's a better way to configure the network. If anyone could provide some advice I would appreciate it. If the setup is reasonable, how do I get Asterisk to reply on eth1? Thanks in advance Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Audio problems on Zap SIP, local network, not IRQ related?
I have made the following additional changes: - enabling MMX extensions in the Asterisk Makefile and remade, installed - disabled parallel, serial, and mouse ports in the BIOS - reenabled ACPI as I was getting errors in the log file The audio problems still exist. Any further advice on how to improve the audio quality would be greatly appreciated. Regards Cameron - Original Message - From: kjcsb [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, June 02, 2006 9:31 PM Subject: Audio problems on Zap SIP, local network, not IRQ related? I am trying to get to the bottom of audio clicks, pops, dropouts with my Asterisk server. These occur even when the system is under minimal load (e.g. 1 Zap device in a queue being played music on hold) and occurs with both Zap and Sip devices so isn't network related. The audio problems occur at the same time on all channels and seems to be when Asterisk gets busy and uses 50% CPU e.g. just after an announcement (you are first in the queue). However running ztspeed (which takes CPU usage to 100%) seems to have no impact on zttest numbers or the audio. I have the following setup: Fedora Core 4 2.6.16-1.2111 smp kernel TDM400 with 2 FXO and 2 FXS modules Various SIP devices SCSI hard drives 2 x P3 processors 500 MHz 1GB RAM Based on what I've read I would have thought that this setup could easily handle one call being played music on hold in a queue. I have read http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html and various other postings to try and resolve this issue. When I run zttest I get 99.987793% most of the time but occasionally it drops to 99.926758% which often corresponds with the audio degradation. I have, however, noticed audio degradation at 99.987793% and good audio at 99.87%. I have followed the instructions on disabling the Linux frame buffer http://www.voip-info.org/wiki/index.php?page=Asterisk+disable+frame+buffer. (There was no graphic on boot and no vga entry in /boot/grub/menu.lst I am not running X windows. The processors are not hyperthreading. I have disabled ACPI by adding acpi=off to /etc/grub.conf cat /proc/interrupts indicates no shared IRQs CPU0 CPU1 0: 21418593 21478150IO-APIC-edge timer 1:142 62IO-APIC-edge i8042 8: 1 0IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 16: 60618 60379 IO-APIC-level megaraid 17:41851894271566 IO-APIC-level aic7xxx, aic7xxx 18:1589117 12 IO-APIC-level eth0 19: 0 0 IO-APIC-level uhci_hcd:usb1 20: 85250811 86274087 IO-APIC-level wctdm NMI: 0 0 LOC: 42898234 42898233 ERR: 0 MIS: 0 lspci -v 00:0a.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b100:0003 Flags: bus master, medium devsel, latency 32, IRQ 20 I/O ports at d800 [size=256] Memory at fe10 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 lspci -vb 00:0a.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b100:0003 Flags: bus master, medium devsel, latency 32, IRQ 11 I/O ports at d800 Memory at fe10 (32-bit, non-prefetchable) Capabilities: [40] Power Management version 2 No other device is using IRQ 20 (or 11) Running zttool shows no alarms, IRQ misses on the TDM400P cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 6 model : 7 model name : Pentium III (Katmai) stepping: 3 cpu MHz : 498.847 cache size : 512 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 2 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 mmx fxsr sse bogomips: 999.41 processor : 1 vendor_id : GenuineIntel cpu family : 6 model : 7 model name : Pentium III (Katmai) stepping: 3 cpu MHz : 498.847 cache size : 512 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 2 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 mmx fxsr sse bogomips: 997.59 Does anyone have any further suggestions? I would really appreciate any other pointers. Regards Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo
[Asterisk-Users] Re: Audio problems on Zap SIP, local network, not IRQ related?
I've read your post on the asterisk mailing list. Agree that the specs of that box should easily handle one call with decent quality. The only thing I can think of right now is to start using the IRQ affinity stuff to move the scsi ethernet modules over to e.g. CPU2 and let the wctdm driver stick to CPU1. You may also want to read the part about setting the latency of the card. Iirc the higher the value the better it is for a Digium card but check it in the docs at http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html or the attached file. I have attempted to change the SMP affinity so that the TDM400 is handled by CPU1: echo 2 /proc/irq/20/smp_affinity cat /proc/irq/20/smp_affinity 0002 However I can see the interrupts for IRQ 20 are still incrementing on both CPUs. Indeed, after about 10 seconds... cat /proc/irq/20/smp_affinity 0001 On further investigation, the SMP affinity on ALL of the IRQs is set to 0001. This implies that everything is handled by CPU0, which it clearly is not! I'll do some more research on this but in the meantime if anyone has any advice on this issue I would appreciate it. I'm not familiar with the megaraid module. You seem to have an adaptec scsi controller. Why need the megaraid module? And if you don't need the usb uhci module, disable usb in the bios. Come to think of it, since you use scsi, disable the IDE ports too. Megaraid is Dell's SCSI hardware RAID controller (this is a Dell Poweredge 2300). I believe the Adaptec stuff is what the Tape, CDROM(!) etc run off. There are no options in the BIOS to disable USB or IDE. Just saw your other posting to the asterisk mailing list. Afaik it is not advisable to use MMX so disable it in the Makefile again and recompile. I don't know which distro you use but if it's CentOS, RHEL or Fedora Core 4 or 5 you can use the rpms at http://laimbock.com/asterisk/ Regards, Patrick I'm running FC4 but some files are patched for New Zealand so I don't think I can use these RPMs. Regards Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Audio problems on Zap SIP, local network, not IRQ related?
I am trying to get to the bottom of audio clicks, pops, dropouts with my Asterisk server. These occur even when the system is under minimal load (e.g. 1 Zap device in a queue being played music on hold) and occurs with both Zap and Sip devices so isn't network related. The audio problems occur at the same time on all channels and seems to be when Asterisk gets busy and uses 50% CPU e.g. just after an announcement (you are first in the queue). However running ztspeed (which takes CPU usage to 100%) seems to have no impact on zttest numbers or the audio. I have the following setup: Fedora Core 4 2.6.16-1.2111 smp kernel TDM400 with 2 FXO and 2 FXS modules Various SIP devices SCSI hard drives 2 x P3 processors 500 MHz 1GB RAM Based on what I've read I would have thought that this setup could easily handle one call being played music on hold in a queue. I have read http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html and various other postings to try and resolve this issue. When I run zttest I get 99.987793% most of the time but occasionally it drops to 99.926758% which often corresponds with the audio degradation. I have, however, noticed audio degradation at 99.987793% and good audio at 99.87%. I have followed the instructions on disabling the Linux frame buffer http://www.voip-info.org/wiki/index.php?page=Asterisk+disable+frame+buffer. (There was no graphic on boot and no vga entry in /boot/grub/menu.lst I am not running X windows. The processors are not hyperthreading. I have disabled ACPI by adding acpi=off to /etc/grub.conf cat /proc/interrupts indicates no shared IRQs CPU0 CPU1 0: 21418593 21478150IO-APIC-edge timer 1:142 62IO-APIC-edge i8042 8: 1 0IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 16: 60618 60379 IO-APIC-level megaraid 17:41851894271566 IO-APIC-level aic7xxx, aic7xxx 18:1589117 12 IO-APIC-level eth0 19: 0 0 IO-APIC-level uhci_hcd:usb1 20: 85250811 86274087 IO-APIC-level wctdm NMI: 0 0 LOC: 42898234 42898233 ERR: 0 MIS: 0 lspci -v 00:0a.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b100:0003 Flags: bus master, medium devsel, latency 32, IRQ 20 I/O ports at d800 [size=256] Memory at fe10 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 lspci -vb 00:0a.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b100:0003 Flags: bus master, medium devsel, latency 32, IRQ 11 I/O ports at d800 Memory at fe10 (32-bit, non-prefetchable) Capabilities: [40] Power Management version 2 No other device is using IRQ 20 (or 11) Running zttool shows no alarms, IRQ misses on the TDM400P cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 6 model : 7 model name : Pentium III (Katmai) stepping: 3 cpu MHz : 498.847 cache size : 512 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 2 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 mmx fxsr sse bogomips: 999.41 processor : 1 vendor_id : GenuineIntel cpu family : 6 model : 7 model name : Pentium III (Katmai) stepping: 3 cpu MHz : 498.847 cache size : 512 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 2 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 mmx fxsr sse bogomips: 997.59 Does anyone have any further suggestions? I would really appreciate any other pointers. Regards Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
