Re: [asterisk-users] Error to install Asterisk
On Tuesday 05 March 2013, termo termosel wrote: Hi, when I try to install Asterisk 11.2.1 the console return error which it tells: /usr/bin/ld: final link failed: No space left on device and the process exits installation. How can I solve this problem? Tmp folder is empty. Thanks,Jordi Try entering this command: # df -h and paste the complete output in a message. This will show the amount of space used and remaining on all filesystems, in human-readable notation (i.e. it will automatically select the units: bytes, kilo, mega, giga or terabytes, so as to get a sensible figure). You'll almost certainly have to move some files out of the way. Have you got, or can you get, a USB external HDD; which either already has a Linux ext4 file system on it, or contains only sacrificial data? -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error to install Asterisk
Hi, Ok, tomorrow I will send the output when I will be in the office! Thanks! From: asterisk_l...@earthshod.co.uk To: asterisk-users@lists.digium.com Date: Tue, 5 Mar 2013 16:11:01 + Subject: Re: [asterisk-users] Error to install Asterisk On Tuesday 05 March 2013, termo termosel wrote: Hi, when I try to install Asterisk 11.2.1 the console return error which it tells: /usr/bin/ld: final link failed: No space left on device and the process exits installation. How can I solve this problem? Tmp folder is empty. Thanks,Jordi Try entering this command: # df -h and paste the complete output in a message. This will show the amount of space used and remaining on all filesystems, in human-readable notation (i.e. it will automatically select the units: bytes, kilo, mega, giga or terabytes, so as to get a sensible figure). You'll almost certainly have to move some files out of the way. Have you got, or can you get, a USB external HDD; which either already has a Linux ext4 file system on it, or contains only sacrificial data? -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ERROR: chan_dahdi.c: PRI Span: 3 PROBLEM: General: Badly Structured Component
Hi Shitian, the line works but the ERROR is annoying since it appears very frequently. I think I'll have to patch it in order to lower its priority, maybe a NOTICE. G On 02/22/2013 03:06 PM, Shitian Long wrote: Did you get it to work may I ask ? On Feb 20, 2013, at 3:49 PM, gincantalupogincantal...@fgasoftware.com wrote: Hi all, has anybody ever encountered this ERROR before? It happens frequently on my debian6-based pbx. I'm using Asterisk 1.8.11 with dahdi-linux-2.4.1 and a quadBRI card. ERROR: chan_dahdi.c: PRI Span: 3PROBLEM: General: Badly Structured Component I tried to google but without success. Do you know what it means? Should I worry? Thank You Giorgio -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ERROR: chan_dahdi.c: PRI Span: 3 PROBLEM: General: Badly Structured Component
Did you get it to work may I ask ? On Feb 20, 2013, at 3:49 PM, gincantalupo gincantal...@fgasoftware.com wrote: Hi all, has anybody ever encountered this ERROR before? It happens frequently on my debian6-based pbx. I'm using Asterisk 1.8.11 with dahdi-linux-2.4.1 and a quadBRI card. ERROR: chan_dahdi.c: PRI Span: 3PROBLEM: General: Badly Structured Component I tried to google but without success. Do you know what it means? Should I worry? Thank You Giorgio -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ERROR: chan_dahdi.c: PRI Span: 3 PROBLEM: General: Badly Structured Component
Hi all, has anybody ever encountered this ERROR before? It happens frequently on my debian6-based pbx. I'm using Asterisk 1.8.11 with dahdi-linux-2.4.1 and a quadBRI card. ERROR: chan_dahdi.c: PRI Span: 3PROBLEM: General: Badly Structured Component I tried to google but without success. Do you know what it means? Should I worry? Thank You Giorgio -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ERROR: chan_dahdi.c: PRI Span: 3 PROBLEM: General: Badly Structured Component
has anybody ever encountered this ERROR before? It happens frequently on my debian6-based pbx. I'm using Asterisk 1.8.11 with dahdi-linux-2.4.1 and a quadBRI card. ERROR: chan_dahdi.c: PRI Span: 3PROBLEM: General: Badly Structured Component I tried to google but without success. Do you know what it means? Should I worry? It means that the peer has rejected a facility message sent by Asterisk. Facility messages are mainly used to implement supplementary services. Supplementary services are things like call-completion, explicit-call-transfer, call-diversion/redirection, and advice-of-charge. The supplementary service that Asterisk was attempting to invoke was rejected and thus failed. It could be that the peer does not support the service, does not recognize the format used, or does not handle the message correctly. A pri set debug on span x trace is needed to give any more information. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error: astobj2.c: refcount -1 on object
On Wed, 2012-11-14 at 09:48 -0800, Michael L. Young wrote: - Original Message - From: Ishfaq Malik i...@pack-net.co.uk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, November 14, 2012 9:25:37 AM Subject: Re: [asterisk-users] Error: astobj2.c: refcount -1 on object Thanks for the advice but that's not really a quick and easy option for us. We would not be able to upgrade to another version without doing full regression testing on the candidate upgrade version and we've been using this version for at least half a year and this is the first time we've had this crash and this error. Also, just upgrading doesn't enlighten me to what is going on to cause this error. Sorry, I thought I was offering a quick and easy option. Since, this is a minor release upgrade, there shouldn't (won't say there won't) be any changes to cause you problems. But, I understand you have to follow your procedures before upgrading. Software is not perfect. Since this is not happening all the time, it may take a while for you to try and figure out what is wrong; that is if you can reproduce it. Therefore, in my opinion, I think you would be better off using your time to consider upgrading especially with a lot of bugs and security updates being in the latest version. Use that time to run through your regression testing. Anyways, if you want to go the path to try and figure out what caused this, I beleive you will need to look at the following information: https://wiki.asterisk.org/wiki/display/AST/Reference+Count+Debugging Hope that helps, Michael (elguero) Hi Sorry if I sounded a bit short with you before. You are correct and we do need to consider upgrading but have been bitten in the past with upgrading, hence having to have a fixed and thorough procedure which usually lasts about a month! It's also a bit trick trying to decide which version to upgrade to as we simply cannot upgrade our production servers every time a new asterisk version comes out. Also, thanks for pointing me in the right direction with the link provided. Regards Ish -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error: astobj2.c: refcount -1 on object
Hi I'm using 1.8.7.0. This morning I got an alert telling me Asterisk on my-host exited on signal 11. Might want to take a peek. When I had a look at the logs I can see a lot of errors like ERROR[14924] astobj2.c: refcount -1 on object 0x2aaad4069068 ERROR[14924] astobj2.c: refcount -2 on object 0x2aaad4069068 All the way up to ERROR[14924] astobj2.c: refcount -2004 on object 0x2aaad4069068 Everything up to this point was completely normal. Does anyone know what this error means and what causes it? Regards Ish -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error: astobj2.c: refcount -1 on object
- Original Message - From: Ishfaq Malik i...@pack-net.co.uk To: asterisk-users@lists.digium.com Sent: Wednesday, November 14, 2012 4:05:21 AM Subject: [asterisk-users] Error: astobj2.c: refcount -1 on object Hi I'm using 1.8.7.0. This morning I got an alert telling me Asterisk on my-host exited on signal 11. Might want to take a peek. When I had a look at the logs I can see a lot of errors like ERROR[14924] astobj2.c: refcount -1 on object 0x2aaad4069068 ERROR[14924] astobj2.c: refcount -2 on object 0x2aaad4069068 All the way up to ERROR[14924] astobj2.c: refcount -2004 on object 0x2aaad4069068 Everything up to this point was completely normal. Does anyone know what this error means and what causes it? I would recommend updating to the latest version. We are up to 1.8.18 and 1.8.19 is around the corner. There have been a lot of bug fixes and you might find that whatever caused this issue is already fixed. Michael (elguero) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error: astobj2.c: refcount -1 on object
On Wed, 2012-11-14 at 05:27 -0800, Michael L. Young wrote: - Original Message - From: Ishfaq Malik i...@pack-net.co.uk To: asterisk-users@lists.digium.com Sent: Wednesday, November 14, 2012 4:05:21 AM Subject: [asterisk-users] Error: astobj2.c: refcount -1 on object Hi I'm using 1.8.7.0. This morning I got an alert telling me Asterisk on my-host exited on signal 11. Might want to take a peek. When I had a look at the logs I can see a lot of errors like ERROR[14924] astobj2.c: refcount -1 on object 0x2aaad4069068 ERROR[14924] astobj2.c: refcount -2 on object 0x2aaad4069068 All the way up to ERROR[14924] astobj2.c: refcount -2004 on object 0x2aaad4069068 Everything up to this point was completely normal. Does anyone know what this error means and what causes it? I would recommend updating to the latest version. We are up to 1.8.18 and 1.8.19 is around the corner. There have been a lot of bug fixes and you might find that whatever caused this issue is already fixed. Michael (elguero) Hi Thanks for the advice but that's not really a quick and easy option for us. We would not be able to upgrade to another version without doing full regression testing on the candidate upgrade version and we've been using this version for at least half a year and this is the first time we've had this crash and this error. Also, just upgrading doesn't enlighten me to what is going on to cause this error. Regards Ish -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error: astobj2.c: refcount -1 on object
- Original Message - From: Ishfaq Malik i...@pack-net.co.uk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, November 14, 2012 9:25:37 AM Subject: Re: [asterisk-users] Error: astobj2.c: refcount -1 on object Thanks for the advice but that's not really a quick and easy option for us. We would not be able to upgrade to another version without doing full regression testing on the candidate upgrade version and we've been using this version for at least half a year and this is the first time we've had this crash and this error. Also, just upgrading doesn't enlighten me to what is going on to cause this error. Sorry, I thought I was offering a quick and easy option. Since, this is a minor release upgrade, there shouldn't (won't say there won't) be any changes to cause you problems. But, I understand you have to follow your procedures before upgrading. Software is not perfect. Since this is not happening all the time, it may take a while for you to try and figure out what is wrong; that is if you can reproduce it. Therefore, in my opinion, I think you would be better off using your time to consider upgrading especially with a lot of bugs and security updates being in the latest version. Use that time to run through your regression testing. Anyways, if you want to go the path to try and figure out what caused this, I beleive you will need to look at the following information: https://wiki.asterisk.org/wiki/display/AST/Reference+Count+Debugging Hope that helps, Michael (elguero) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error SIP/2.0 488 Not acceptable here
Hello, a person trying to call me by my phone number is getting the error 488 Not acceptable here. I googled that error, seems like this error is normally caused by a failed codec negotation, though I have no clue how I could have read this out of the logs. Anyway, my setup is as follows: Asterisk 1.8.13.0 - NAT - Sipgate SIP Provider The user calling me is also using Sipgate and is calling my landline phone number from Sipgate (not [my sip id]@sipgate.de). My sip.conf including the codec restrictions looks like this (I left out my local sip account) [general] port=5060 bindaddr=0.0.0.0 context=other language=de allowguest=no qualify=no disallow=all allow=alaw allow=ulaw allow=g729 allow=gsm allow=slinear srvlookup=yes register = MY_SIP_ID:password@sipgate.de/MY_SIP_ID [sipgate] type=friend insecure=invite nat=yes username=MY_SIP_ID fromuser=MY_SIP_ID fromdomain=sipgate.de secret=password host=sipgate.de qualify=yes canreinvite=no dtmfmode=rfc2833 context = from_external_voip_provider The relevant part from my full asterisk log /var/log/asterisk/full including the 488 Not acceptable here error message: [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: --- SIP read from UDP:217.10.79.9:5060 --- INVITE sip:MY_SIP_ID@192.168.5.11:5060 SIP/2.0 Record-Route: sip:217.10.79.9;lr;ftag=8cgn1bajqb Record-Route: sip:172.20.40.3;lr=on Record-Route: sip:217.10.79.9;lr;ftag=8cgn1bajqb Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK8f5c.48627b3.0 Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bK8f5c.48627b3.0 Via: SIP/2.0/UDP 217.10.79.9:5060 ;received=217.10.68.222;branch=z9hG4bK-un6p0cm50qse Via: SIP/2.0/UDP 192.168.0.8:2048 ;received=CALLING_PARTY_IP_ADDRESS;branch=z9hG4bK-un6p0cm50qse;rport=2048 From: sipgate.de sip:CALLING_PARTY_SIP_ID@sipgate.de;tag=8cgn1bajqb To: sip:0049MY_PHONE_NUMBER@sipgate.de;user=phone Call-ID: 4fdf703d880d-ywqwnfbbj1h7 CSeq: 2 INVITE Max-Forwards: 67 Contact: sip:CALLING_PARTY_SIP_ID@CALLING_PARTY_IP_ADDRESS:2048;line=swnt2d3t;reg-id=1 X-Serialnumber: 000413251D76 User-Agent: snom300/8.7.3.7 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 522 P-Asserted-Identity: sip:CALLING_PARTY_PHONE_NUMBER@sipgate.de v=0 o=root 269390684 269390684 IN IP4 192.168.0.8 s=call c=IN IP4 217.10.77.20 t=0 0 m=audio 62652 RTP/AVP 9 0 8 3 99 108 18 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:Ed8iHaP3BXNVeXHj98PRa6sJyImNer3ImjUvDZps a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:108 AAL2-G726-32/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=direction:active a=nortpproxy:yes - [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: --- (25 headers 21 lines) --- [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Sending to 217.10.79.9:5060(NAT) [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Using INVITE request as basis request - 4fdf703d880d-ywqwnfbbj1h7 [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found peer 'sipgate' for 'CALLING_PARTY_SIP_ID' from 217.10.79.9:5060 [Jun 18 20:15:26] VERBOSE[1164] netsock2.c: == Using SIP RTP CoS mark 5 [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format 9 [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format 0 [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format 8 [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format 3 [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format 99 [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format 108 [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format 18 [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format 101 [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description format G722 for ID 9 [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description format PCMU for ID 0 [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description format PCMA for ID 8 [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description format GSM for ID 3 [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description format G726-32 for ID 99 [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description format AAL2-G726-32 for ID 108 [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description format G729 for ID 18 [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description format telephone-event for ID 101 [Jun 18 20:15:26] WARNING[1164] chan_sip.c: We are requesting SRTP, but they responded without it! [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: --- Reliably Transmitting (NAT) to 217.10.79.9:5060 ---
Re: [asterisk-users] Error SIP/2.0 488 Not acceptable here
- Original Message - From: Stefan at WPF stefan.at@googlemail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, June 18, 2012 3:04:32 PM Subject: [asterisk-users] Error SIP/2.0 488 Not acceptable here Hello, a person trying to call me by my phone number is getting the error 488 Not acceptable here. I googled that error, seems like this error is normally caused by a failed codec negotation, though I have no clue how I could have read this out of the logs. Anyway, my setup is as follows: Asterisk 1.8.13.0 - NAT - Sipgate SIP Provider The user calling me is also using Sipgate and is calling my landline phone number from Sipgate (not [my sip id]@ sipgate.de ). My sip.conf including the codec restrictions looks like this (I left out my local sip account) [general] port=5060 bindaddr=0.0.0.0 context=other language=de allowguest=no qualify=no disallow=all allow=alaw allow=ulaw allow=g729 allow=gsm allow=slinear srvlookup=yes register = MY_SIP_ID:password@ sipgate.de/ MY_SIP_ID [sipgate] type=friend insecure=invite nat=yes username=MY_SIP_ID fromuser=MY_SIP_ID fromdomain= sipgate.de secret=password host= sipgate.de qualify=yes canreinvite=no dtmfmode=rfc2833 context = from_external_voip_provider The relevant part from my full asterisk log /var/log/asterisk/full including the 488 Not acceptable here error message: [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: --- SIP read from UDP: 217.10.79.9:5060 --- INVITE sip:MY_SIP_ID@ 192.168.5.11:5060 SIP/2.0 Record-Route: sip:217.10.79.9;lr;ftag=8cgn1bajqb Record-Route: sip:172.20.40.3;lr=on Record-Route: sip:217.10.79.9;lr;ftag=8cgn1bajqb Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK8f5c.48627b3.0 Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bK8f5c.48627b3.0 Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK-un6p0cm50qse Via: SIP/2.0/UDP 192.168.0.8:2048;received=CALLING_PARTY_IP_ADDRESS;branch=z9hG4bK-un6p0cm50qse;rport=2048 From: sipgate.de sip:CALLING_PARTY_SIP_ID@ sipgate.de ;tag=8cgn1bajqb To: sip:0049MY_PHONE_NUMBER@ sipgate.de ;user=phone Call-ID: 4fdf703d880d-ywqwnfbbj1h7 CSeq: 2 INVITE Max-Forwards: 67 Contact: sip:CALLING_PARTY_SIP_ID@CALLING_PARTY_IP_ADDRESS:2048;line=swnt2d3t;reg-id=1 X-Serialnumber: 000413251D76 User-Agent: snom300/ 8.7.3.7 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 522 P-Asserted-Identity: sip:CALLING_PARTY_PHONE_NUMBER@ sipgate.de v=0 o=root 269390684 269390684 IN IP4 192.168.0.8 s=call c=IN IP4 217.10.77.20 t=0 0 m=audio 62652 RTP/AVP 9 0 8 3 99 108 18 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:Ed8iHaP3BXNVeXHj98PRa6sJyImNer3ImjUvDZps a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:108 AAL2-G726-32/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=direction:active a=nortpproxy:yes - [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: --- (25 headers 21 lines) --- [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Sending to 217.10.79.9:5060 (NAT) [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Using INVITE request as basis request - 4fdf703d880d-ywqwnfbbj1h7 [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found peer 'sipgate' for 'CALLING_PARTY_SIP_ID' from 217.10.79.9:5060 [Jun 18 20:15:26] VERBOSE[1164] netsock2.c: == Using SIP RTP CoS mark 5 [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format 9 [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format 0 [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format 8 [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format 3 [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format 99 [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format 108 [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format 18 [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format 101 [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description format G722 for ID 9 [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description format PCMU for ID 0 [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description format PCMA for ID 8 [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio
Re: [asterisk-users] Error SIP/2.0 488 Not acceptable here
Matthew, thank you very much for the fast reply and very likely the solution! Using your hint I could locally reproduce the 488 Not Acceptable on my Snom 300 with FW 8.7.3.7 by setting RTP encryption (SRTP) to on and RTP/SAVP to off (RTP/SAVP mandatory or SRTP off - all fine). The person calling me has the same phone and FW, so that should really be the problem, I will tell him to change it. However I am wondering why it's possible to configure a Snom phone in such a wrong way at all? Is that necessary for some legacy systems? Also I am wondering if it's possible to tell asterisk to just ignore the crypto line when the profile is just RTP/AVP / not to take things that serious? In this specific case I can tell the calling person to change the setting, but unfortunately I can't tell this every person calling me (not only because they simply can't call me). 2012/6/18 Matthew Jordan mjor...@digium.com - Original Message - From: Stefan at WPF stefan.at@googlemail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, June 18, 2012 3:04:32 PM Subject: [asterisk-users] Error SIP/2.0 488 Not acceptable here Hello, a person trying to call me by my phone number is getting the error 488 Not acceptable here. I googled that error, seems like this error is normally caused by a failed codec negotation, though I have no clue how I could have read this out of the logs. Anyway, my setup is as follows: Asterisk 1.8.13.0 - NAT - Sipgate SIP Provider The user calling me is also using Sipgate and is calling my landline phone number from Sipgate (not [my sip id]@ sipgate.de ). My sip.conf including the codec restrictions looks like this (I left out my local sip account) [general] port=5060 bindaddr=0.0.0.0 context=other language=de allowguest=no qualify=no disallow=all allow=alaw allow=ulaw allow=g729 allow=gsm allow=slinear srvlookup=yes register = MY_SIP_ID:password@ sipgate.de/ MY_SIP_ID [sipgate] type=friend insecure=invite nat=yes username=MY_SIP_ID fromuser=MY_SIP_ID fromdomain= sipgate.de secret=password host= sipgate.de qualify=yes canreinvite=no dtmfmode=rfc2833 context = from_external_voip_provider The relevant part from my full asterisk log /var/log/asterisk/full including the 488 Not acceptable here error message: [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: --- SIP read from UDP: 217.10.79.9:5060 --- INVITE sip:MY_SIP_ID@ 192.168.5.11:5060 SIP/2.0 Record-Route: sip:217.10.79.9;lr;ftag=8cgn1bajqb Record-Route: sip:172.20.40.3;lr=on Record-Route: sip:217.10.79.9;lr;ftag=8cgn1bajqb Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK8f5c.48627b3.0 Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bK8f5c.48627b3.0 Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK-un6p0cm50qse Via: SIP/2.0/UDP 192.168.0.8:2048 ;received=CALLING_PARTY_IP_ADDRESS;branch=z9hG4bK-un6p0cm50qse;rport=2048 From: sipgate.de sip:CALLING_PARTY_SIP_ID@ sipgate.de ;tag=8cgn1bajqb To: sip:0049MY_PHONE_NUMBER@ sipgate.de ;user=phone Call-ID: 4fdf703d880d-ywqwnfbbj1h7 CSeq: 2 INVITE Max-Forwards: 67 Contact: sip:CALLING_PARTY_SIP_ID@CALLING_PARTY_IP_ADDRESS:2048;line=swnt2d3t;reg-id=1 X-Serialnumber: 000413251D76 User-Agent: snom300/ 8.7.3.7 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 522 P-Asserted-Identity: sip:CALLING_PARTY_PHONE_NUMBER@ sipgate.de v=0 o=root 269390684 269390684 IN IP4 192.168.0.8 s=call c=IN IP4 217.10.77.20 t=0 0 m=audio 62652 RTP/AVP 9 0 8 3 99 108 18 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:Ed8iHaP3BXNVeXHj98PRa6sJyImNer3ImjUvDZps a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:108 AAL2-G726-32/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=direction:active a=nortpproxy:yes - [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: --- (25 headers 21 lines) --- [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Sending to 217.10.79.9:5060 (NAT) [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Using INVITE request as basis request - 4fdf703d880d-ywqwnfbbj1h7 [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found peer 'sipgate' for 'CALLING_PARTY_SIP_ID' from 217.10.79.9:5060
Re: [asterisk-users] Error SIP/2.0 488 Not acceptable here
- Original Message - From: Stefan at WPF stefan.at@googlemail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, June 18, 2012 4:05:04 PM Subject: Re: [asterisk-users] Error SIP/2.0 488 Not acceptable here Matthew, thank you very much for the fast reply and very likely the solution! Using your hint I could locally reproduce the 488 Not Acceptable on my Snom 300 with FW 8.7.3.7 by setting RTP encryption (SRTP) to on and RTP/SAVP to off (RTP/SAVP mandatory or SRTP off - all fine). The person calling me has the same phone and FW, so that should really be the problem, I will tell him to change it. However I am wondering why it's possible to configure a Snom phone in such a wrong way at all? Is that necessary for some legacy systems? I'm not sure why it would be a configuration option. The 'must' I referred to comes from Section 6 of RFC 4568. Its certainly possible that there is some PBX out there that did not understand a SRTP transport designation. Also I am wondering if it's possible to tell asterisk to just ignore the crypto line when the profile is just RTP/AVP / not to take things that serious? In this specific case I can tell the calling person to change the setting, but unfortunately I can't tell this every person calling me (not only because they simply can't call me). Asterisk does not currently have a setting for that - if it encounters a security descriptor in an SDP offer, it will attempt to process it. -- Matthew Jordan Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error about codecs AMR-NB.
Hi. Anyone know how to fix this problem below. I'm add codecs AMR-NB and AMR-WB and I get error in AMR-NB about this Found unknown media description format AMR for ID, a search about this on google and I can't find any solution about this. Thanks in advanced and best regards. Julio Lemos (8 headers 0 lines) --- --- SIP read from UDP:192.168.3.227:45327 ---INVITE sip:5432@192.168.3.148 SIP/2.0Via: SIP/2.0/UDP 192.168.3.227:45327;rport;branch=z9hG4bKPjQv3ZDrKzwWFMfOO4eFG0AECvwqEerQ9AMax-Forwards: 70From: 5505 sip:5505@192.168.3.148;tag=H24FA8xsH7K9DpuES9CbWgGkvbYSfF8qTo: sip:5432@192.168.3.148Contact: 5505 sip:5505@192.168.3.227:45327;obCall-ID: WyL8a33LsNOaQjw2WbVUN7qj0R4hyEMOCSeq: 13600 INVITEAllow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONSSupported: replaces, 100rel, timer, norefersubSession-Expires: 1800Min-SE: 90User-Agent: CSipSimple r1108 / umts_milestone2-10Authorization: Digest username=5505, realm=asterisk, nonce=52b8e2d3, uri=sip:5432@192.168.3.148, response=1a95c0bbb6f5143037eb6f7b6f2f6674, algorithm=MD5Content-Type: application/sdpContent-Length: 305 v=0o=- 3547457675 3547457675 IN IP4 192.168.3.227s=pjmediac=IN IP4 192.168.3.227t=0 0a=X-nat:0m=audio 4000 RTP/AVP 108 8 0 101a=rtcp:4001 IN IP4 192.168.3.227a=rtpmap:108 AMR/8000a=rtpmap:8 PCMA/8000a=rtpmap:0 PCMU/8000a=sendrecva=rtpmap:101 telephone-event/8000a=fmtp:101 0-15 (16 headers 14 lines) ---Sending to 192.168.3.227:45327 (NAT)Using INVITE request as basis request - WyL8a33LsNOaQjw2WbVUN7qj0R4hyEMOFound peer '5505' for '5505' from 192.168.3.227:45327 == Using SIP RTP CoS mark 5Found RTP audio format 108Found RTP audio format 8Found RTP audio format 0Found RTP audio format 101Found unknown media description format AMR for ID 108Found audio description format PCMA for ID 8Found audio description format PCMU for ID 0Found audio description format telephone-event for ID 101[May 31 09:55:05] NOTICE[28286]: chan_sip.c:9372 process_sdp: No compatible codecs, not accepting this offer! --- Reliably Transmitting (NAT) to 192.168.3.227:45327 ---SIP/2.0 488 Not acceptable hereVia: SIP/2.0/UDP 192.168.3.227:45327;branch=z9hG4bKPjQv3ZDrKzwWFMfOO4eFG0AECvwqEerQ9A;received=192.168.3.227;rport=45327From: 5505 sip:5505@192.168.3.148;tag=H24FA8xsH7K9DpuES9CbWgGkvbYSfF8qTo: sip:5432@192.168.3.148;tag=as02c2bc10Call-ID: WyL8a33LsNOaQjw2WbVUN7qj0R4hyEMOCSeq: 13600 INVITEServer: Asterisk PBX 10.4.0Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISHSupported: replaces, timerContent-Length: 0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] error during dahdi installation on centos
Hello, # rpm -qa | grep kernel kernel-headers-2.6.18-274.18.1.el5 kernel-PAE-2.6.18-128.el5 kernel-devel-2.6.18-274.18.1.el5 kernel-PAE-devel-2.6.18-274.18.1.el5 [root@localhost ~]# uname -i i386 Trying to install dahdi-linux-complete-2.3.0.1+2.3.0 on CentOS but get below error. Can you please assist in this? [root@localhost dahdi-linux-complete-2.3.0.1+2.3.0]# make make -C linux all make[1]: Entering directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux' make -C drivers/dahdi/firmware firmware-loaders make[2]: Entering directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware' make[2]: Leaving directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware' You do not appear to have the sources for the 2.6.18-128.el5PAE kernel installed. make[1]: *** [modules] Error 1 make[1]: Leaving directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux' make: *** [all] Error 2 Thanks, Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error during dahdi installation on centos
On Wed, Feb 15, 2012 at 12:11:18PM +, Kamlesh Kumar wrote: Hello, # rpm -qa | grep kernel kernel-headers-2.6.18-274.18.1.el5 kernel-PAE-2.6.18-128.el5 kernel-devel-2.6.18-274.18.1.el5 kernel-PAE-devel-2.6.18-274.18.1.el5 [root@localhost ~]# uname -i i386 Trying to install dahdi-linux-complete-2.3.0.1+2.3.0 on CentOS but get below error. Can you please assist in this? [root@localhost dahdi-linux-complete-2.3.0.1+2.3.0]# make make -C linux all make[1]: Entering directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux' make -C drivers/dahdi/firmware firmware-loaders make[2]: Entering directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware' make[2]: Leaving directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware' You do not appear to have the sources for the 2.6.18-128.el5PAE kernel installed. make[1]: *** [modules] Error 1 make[1]: Leaving directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux' make: *** [all] Error 2 Boot to the newer kernel and/or use: make KVERS=2.6.18-274.18.1.el5PAE -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error during dahdi installation on centos
thank you very much for your quick response. make KVERS=2.6.18-274.18.1.el5PAE It started the installation but stuck at below error LD [M] /usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/wcte12xp/wcte12xp.o CC [M] /usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/xpp/card_bri.o In file included from /usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/xpp/xpd.h:31, from /usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/xpp/card_bri.c:29: include/linux/device.h:407: error: expected identifier or â(â before âconstâ make[4]: *** [/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/xpp/card_bri.o] Error 1 make[3]: *** [/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/xpp] Error 2 make[2]: *** [_module_/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi] Error 2 make[2]: Leaving directory `/usr/src/kernels/2.6.18-274.18.1.el5-PAE-i686' make[1]: *** [modules] Error 2 make[1]: Leaving directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux' make: *** [all] Error 2 Regards, Kamlesh Date: Wed, 15 Feb 2012 14:39:00 +0200 From: tzafrir.co...@xorcom.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] error during dahdi installation on centos On Wed, Feb 15, 2012 at 12:11:18PM +, Kamlesh Kumar wrote: Hello, # rpm -qa | grep kernel kernel-headers-2.6.18-274.18.1.el5 kernel-PAE-2.6.18-128.el5 kernel-devel-2.6.18-274.18.1.el5 kernel-PAE-devel-2.6.18-274.18.1.el5 [root@localhost ~]# uname -i i386 Trying to install dahdi-linux-complete-2.3.0.1+2.3.0 on CentOS but get below error. Can you please assist in this? [root@localhost dahdi-linux-complete-2.3.0.1+2.3.0]# make make -C linux all make[1]: Entering directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux' make -C drivers/dahdi/firmware firmware-loaders make[2]: Entering directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware' make[2]: Leaving directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware' You do not appear to have the sources for the 2.6.18-128.el5PAE kernel installed. make[1]: *** [modules] Error 1 make[1]: Leaving directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux' make: *** [all] Error 2 Boot to the newer kernel and/or use: make KVERS=2.6.18-274.18.1.el5PAE -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error: Autodestruct on dialog
$ds4rưdeseiiijp Sent from my Sony Ericsson Xperia neo Kevin P. Fleming kpflem...@digium.com wrote: On 08/05/2011 09:43 AM, Christian Pinedo Zamalloa wrote: Hi all, I need to wait several seconds in h extension. Since Wait application doesn't work in h extension I must use System in the following way: exten = h,1, same = n,... same = n,System(/bin/sleep 25) same = n,... But when I use this System command in h extension I get the following warning: [Aug 5 14:19:50] WARNING[23637] chan_sip.c: Autodestruct on dialog '7249D00-BE9611E0-A8B6C958-F31290CD@10.180.4.1' with owner in place (Method: BYE) You are stopping the Asterisk SIP channel driver from doing its job; it expects the channel to be dead much sooner than 25 seconds after receiving (or sending) a BYE. Why do you need to keep the channel alive for so long after it has been hungup? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error: Autodestruct on dialog
I am trying to do a dynamic wrapuptime for a queue. Instead of having to wait always x seconds once an angent has attended succesfully a call, I prefer to give the agent the option to disable his wrapuptime. exten = h,1,PauseQueuemember same = n,System(/bin/sleep 25) same = n,UnpauseQueueMember Now the agent is paused automatically 25 seconds (or whatever I want) after attending succesfully a call and If the agent finishes earlier his administrative work he can unpause his telephone by calling to one extension. exten = 1234,1,UnpauseQueueMember So, I must think I should not do this way??? xD 2011/8/5 Kevin P. Fleming kpflem...@digium.com: On 08/05/2011 09:43 AM, Christian Pinedo Zamalloa wrote: Hi all, I need to wait several seconds in h extension. Since Wait application doesn't work in h extension I must use System in the following way: exten = h,1, same = n,... same = n,System(/bin/sleep 25) same = n,... But when I use this System command in h extension I get the following warning: [Aug 5 14:19:50] WARNING[23637] chan_sip.c: Autodestruct on dialog '7249D00-BE9611E0-A8B6C958-F31290CD@10.180.4.1' with owner in place (Method: BYE) You are stopping the Asterisk SIP channel driver from doing its job; it expects the channel to be dead much sooner than 25 seconds after receiving (or sending) a BYE. Why do you need to keep the channel alive for so long after it has been hungup? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Christian Pinedo Zamalloa (zako) PGP keyID: 0x828D0C80 Fingerprint: 7BFF 4105 F46B 7977 BD96 348C 1007 4FF8 828D 0C80 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] error: Autodestruct on dialog
Hi all, I need to wait several seconds in h extension. Since Wait application doesn't work in h extension I must use System in the following way: exten = h,1, same = n,... same = n,System(/bin/sleep 25) same = n,... But when I use this System command in h extension I get the following warning: [Aug 5 14:19:50] WARNING[23637] chan_sip.c: Autodestruct on dialog '7249D00-BE9611E0-A8B6C958-F31290CD@10.180.4.1' with owner in place (Method: BYE) If i run in the CLI sip show channels there are a lot of SIP dialogs that haven't finished yet and that are hold by Asterisk: asterisk*CLI sip show channels 10.180.4.1 652 5648af9721df9cc 0x0 (nothing)No Rx: BYEcme01 10.180.4.1 650 E546BE0-BEA411E 0x0 (nothing)No Rx: BYEcme01 10.180.4.1 699095244BAC5BF87-BC2811 0x0 (nothing)No Rx: BYEcme01 636 active SIP dialogs But they aren't active channels: asterisk*CLI core show channels Channel Location State Application(Data) 0 active channels 0 active calls 1013 calls processed Could this be a bug or am I doing something bad??? Thanks, -- Christian Pinedo Zamalloa (zako) PGP keyID: 0x828D0C80 Fingerprint: 7BFF 4105 F46B 7977 BD96 348C 1007 4FF8 828D 0C80 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error: Autodestruct on dialog
On 08/05/2011 09:43 AM, Christian Pinedo Zamalloa wrote: Hi all, I need to wait several seconds in h extension. Since Wait application doesn't work in h extension I must use System in the following way: exten = h,1, same = n,... same = n,System(/bin/sleep 25) same = n,... But when I use this System command in h extension I get the following warning: [Aug 5 14:19:50] WARNING[23637] chan_sip.c: Autodestruct on dialog '7249D00-BE9611E0-A8B6C958-F31290CD@10.180.4.1' with owner in place (Method: BYE) You are stopping the Asterisk SIP channel driver from doing its job; it expects the channel to be dead much sooner than 25 seconds after receiving (or sending) a BYE. Why do you need to keep the channel alive for so long after it has been hungup? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error Code 101
Hi, I have 8 port PRI Sangoma Card connected to the Server running Asterisk 1.8.5 on CentOS 5.6. I see these errors in sangoma file under /var/log/asterisk/ http://pastebin.ubuntu.com/651451/ Error code 101 is Message not compatible with call state. The explanation for this is The remote equipment received an unexpected message that does not correspond to the current state of the connection. This is usually due to a D-channel error. Please suggest/guide and let me know if anyone needs any information about configs. Regards, Kaushal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error Code 101
On Mon, Jul 25, 2011 at 4:39 AM, Kaushal Shriyan kaushalshri...@gmail.com wrote: Hi, I have 8 port PRI Sangoma Card connected to the Server running Asterisk 1.8.5 on CentOS 5.6. I see these errors in sangoma file under /var/log/asterisk/ http://pastebin.ubuntu.com/651451/ Error code 101 is Message not compatible with call state. The explanation for this is The remote equipment received an unexpected message that does not correspond to the current state of the connection. This is usually due to a D-channel error. Please suggest/guide and let me know if anyone needs any information about configs. Regards, Kaushal Hi, The versions are as below :- asterisk18.x86_64 1.8.5.0-1_centos5 libpri-1.4.11.5-1_centos5.x86_64 WANPIPE Release: 3.5.20 Regards, Kaushal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error Code 101
On Mon, Jul 25, 2011 at 5:08 AM, Kaushal Shriyan kaushalshri...@gmail.com wrote: On Mon, Jul 25, 2011 at 4:39 AM, Kaushal Shriyan kaushalshri...@gmail.com wrote: Hi, I have 8 port PRI Sangoma Card connected to the Server running Asterisk 1.8.5 on CentOS 5.6. I see these errors in sangoma file under /var/log/asterisk/ http://pastebin.ubuntu.com/651451/ Error code 101 is Message not compatible with call state. The explanation for this is The remote equipment received an unexpected message that does not correspond to the current state of the connection. This is usually due to a D-channel error. Please suggest/guide and let me know if anyone needs any information about configs. Regards, Kaushal Hi, The versions are as below :- asterisk18.x86_64 1.8.5.0-1_centos5 libpri-1.4.11.5-1_centos5.x86_64 WANPIPE Release: 3.5.20 Regards, Kaushal Hi, I have Package libpri-1.4.11.5-1_centos5.x86_64 already installed and latest version on CentOS 5.6, is there a rpm version of 1.4.12 for CentOS 5.6 as per http://downloads.asterisk.org/pub/telephony/libpri/ ? Regards, Kaushal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] error in GUI access
I have this error after upgrading to 1.8.4.4 on my centos 5.6 32it When using GUI to access, I got this error *** glibc detected *** /usr/sbin/asterisk: double free or corruption (!prev): 0x0919c070 *** The server cannot be connected via GUI and the asterisk CLI dropped and exit into linux command line. Appreciate if help can be provided CK -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error in GUI access
On Friday 01 Jul 2011, asterisk asterisk wrote: I have this error after upgrading to 1.8.4.4 on my centos 5.6 32it When using GUI to access, I got this error *** glibc detected *** /usr/sbin/asterisk: double free or corruption (!prev): 0x0919c070 *** The server cannot be connected via GUI and the asterisk CLI dropped and exit into linux command line. Ooo-er. Last time I got an error like this, it turned out that the box had been compromised with a rootkit. Luckily, most rootkits give themselves away in trying to make themselves hard to detect / remove: first they replace some system utilities (which, on Debian, also breaks colour directory listings) with specially munged ones (for instance, an ls command that will deliberately not show any of the rootkit's own extra files; a ps that will not show the extra processes; a netstat that will not show the rootkit's network connections; and so forth) and then they set the extended attributes on the new files to prevent them from being overwritten. So checking extended attributes can give you a clue that all is not well. Try # lsattr /bin # lsattr /usr/bin # lsattr /sbin # lsattr /usr/sbin All files should have a row of - signs in the left hand column. Any a or i in a file's attributes indicates that the file has had its extended attributes modified, and you should be suspicious. Note: ignore any errors such as lsattr: Operation not supported While reading flags on /bin/nc (this just means the file is a symbolic link, and these don't have extended attributes). -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error in GUI access
Hi, I did not find any file with a or i with your suggested commands. Any other clues? CK On Fri, Jul 1, 2011 at 6:23 PM, A J Stiles asterisk_l...@earthshod.co.ukwrote: On Friday 01 Jul 2011, asterisk asterisk wrote: I have this error after upgrading to 1.8.4.4 on my centos 5.6 32it When using GUI to access, I got this error *** glibc detected *** /usr/sbin/asterisk: double free or corruption (!prev): 0x0919c070 *** The server cannot be connected via GUI and the asterisk CLI dropped and exit into linux command line. Ooo-er. Last time I got an error like this, it turned out that the box had been compromised with a rootkit. Luckily, most rootkits give themselves away in trying to make themselves hard to detect / remove: first they replace some system utilities (which, on Debian, also breaks colour directory listings) with specially munged ones (for instance, an ls command that will deliberately not show any of the rootkit's own extra files; a ps that will not show the extra processes; a netstat that will not show the rootkit's network connections; and so forth) and then they set the extended attributes on the new files to prevent them from being overwritten. So checking extended attributes can give you a clue that all is not well. Try # lsattr /bin # lsattr /usr/bin # lsattr /sbin # lsattr /usr/sbin All files should have a row of - signs in the left hand column. Any a or i in a file's attributes indicates that the file has had its extended attributes modified, and you should be suspicious. Note: ignore any errors such as lsattr: Operation not supported While reading flags on /bin/nc (this just means the file is a symbolic link, and these don't have extended attributes). -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error loading module 'res_fax_digium.so'
Hi, I installed Free FAX for Asterisk on my home Asterisk 1.8(Centos5.5) server. Everything seems fine but I just saw this WARNING shows up in the log every time I start the asterisk: /[2011-03-07 10:50:41] WARNING[13429] loader.c: Error loading module 'res_fax_digium.so': /usr/lib/asterisk/modules/res_fax_digium.so: undefined symbol: ast_fax_tech_unregister/ And in later in the log file, I also saw: /[2011-03-07 10:50:43] NOTICE[13429] res_fax_digium.c: Digium FAX technology module version 1.8.0_1.3.0, Copyright (C) 2008-2009 Digium, Inc. [2011-03-07 10:50:43] NOTICE[13429] res_fax_digium.c: This module is supplied under a commercial license granted by Digium, Inc. [2011-03-07 10:50:43] NOTICE[13429] res_fax_digium.c: Please see the full license text supplied by the accompanying [2011-03-07 10:50:43] NOTICE[13429] res_fax_digium.c: register utility, or ask for a copy from Digium. [2011-03-07 10:50:43] NOTICE[13429] res_fax_digium.c: This product includes software developed by the OpenSSL Project [2011-03-07 10:50:43] NOTICE[13429] res_fax_digium.c: for use in the OpenSSL Toolkit. (http://www.openssl.org/) [2011-03-07 10:50:43] NOTICE[13429] res_fax_digium.c: Copyright (C) 1998-2008 The OpenSSL Project/ How can I fix this WARNING error? Thanks. Jian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error loading module 'res_fax_digium.so'
On 03/07/2011 12:58 PM, Jian Gao wrote: Hi, I installed Free FAX for Asterisk on my home Asterisk 1.8(Centos5.5) server. Everything seems fine but I just saw this WARNING shows up in the log every time I start the asterisk: /[2011-03-07 10:50:41] WARNING[13429] loader.c: Error loading module 'res_fax_digium.so': /usr/lib/asterisk/modules/res_fax_digium.so: undefined symbol: ast_fax_tech_unregister/ And in later in the log file, I also saw: /[2011-03-07 10:50:43] NOTICE[13429] res_fax_digium.c: Digium FAX technology module version 1.8.0_1.3.0, Copyright (C) 2008-2009 Digium, Inc. [2011-03-07 10:50:43] NOTICE[13429] res_fax_digium.c: This module is supplied under a commercial license granted by Digium, Inc. [2011-03-07 10:50:43] NOTICE[13429] res_fax_digium.c: Please see the full license text supplied by the accompanying [2011-03-07 10:50:43] NOTICE[13429] res_fax_digium.c: register utility, or ask for a copy from Digium. [2011-03-07 10:50:43] NOTICE[13429] res_fax_digium.c: This product includes software developed by the OpenSSL Project [2011-03-07 10:50:43] NOTICE[13429] res_fax_digium.c: for use in the OpenSSL Toolkit. (http://www.openssl.org/) [2011-03-07 10:50:43] NOTICE[13429] res_fax_digium.c: Copyright (C) 1998-2008 The OpenSSL Project/ How can I fix this WARNING error? You can follow the instructions with the product and ensure that res_fax.so is loaded before res_fax_digium.so. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error compiling Asterisk 1.8.3 on Sun SPARC x64 w/Debian Squeeze
Hello all, Figured I'd repost this with an edited subject line, to attract attention of people with Debian On Sparc experience. Apologies in advance if this kind of thing is frowned upon :) [AR] hash/hash.o hash/hash_bigkey.o hash/hash_buf.o hash/hash_func.o hash/hash_log2.o hash/hash_page.o hash/ndbm.o btree/bt_close.o btree/bt_conv.o btree/bt_debug.o btree/bt_delete.o btree/bt_get.o btree/bt_open.o btree/bt_overflow.o btree/bt_page.o btree/bt_put.o btree/bt_search.o btree/bt_seq.o btree/bt_split.o btree/bt_utils.o db/db.o mpool/mpool.o recno/rec_close.o recno/rec_delete.o recno/rec_get.o recno/rec_open.o recno/rec_put.o recno/rec_search.o recno/rec_seq.o recno/rec_utils.o - libdb1.a [LD] abstract_jb.o acl.o alaw.o aoc.o app.o ast_expr2.o ast_expr2f.o asterisk.o astfd.o astmm.o astobj2.o audiohook.o autochan.o autoservice.o bridging.o callerid.o ccss.o cdr.o cel.o channel.o chanvars.o cli.o config.o data.o datastore.o db.o devicestate.o dial.o dns.o dnsmgr.o dsp.o enum.o event.o features.o file.o fixedjitterbuf.o frame.o framehook.o fskmodem.o global_datastores.o hashtab.o heap.o http.o image.o indications.o io.o jitterbuf.o loader.o lock.o logger.o manager.o md5.o netsock.o netsock2.o pbx.o plc.o poll.o privacy.o rtp_engine.o say.o sched.o security_events.o sha1.o slinfactory.o srv.o ssl.o stdtime/localtime.o strcompat.o strings.o stun.o syslog.o taskprocessor.o tcptls.o tdd.o term.o test.o threadstorage.o timing.o translate.o udptl.o ulaw.o utils.o version.o xml.o xmldoc.o editline/libedit.a db1-ast/libdb1.a - asterisk astobj2.o: In function `ast_atomic_fetchadd_int': /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' astobj2.o:/usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: more undefined references to `__sync_fetch_and_add_4' follow utils.o: In function `ast_atomic_dec_and_test': /usr/src/asterisk-1.8.3/include/asterisk/lock.h:635: undefined reference to `__sync_sub_and_fetch_4' utils.o: In function `ast_atomic_fetchadd_int': /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' collect2: ld returned 1 exit status make[1]: *** [asterisk] Error 1 make: *** [main] Error 2 Any idea where this is coming from? seems like something is selected that doesn't have other related stuff unselected? no clue where to start looking Thanks \RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error compiling Asterisk 1.8.3 on Sun SPARC x64 w/Debian Squeeze
On 03/08/11 08:49, RR wrote: Any idea where this is coming from? seems like something is selected that doesn't have other related stuff unselected? no clue where to start looking No SPARC expert, but I seem to recall the lowest-common-denominator SPARCs lack things like hardware multiply in the instruction set. Even if it doesn't help fix the problem, you probably will want to use at least -mcpu=v9 (educated guess looking at the gcc manpage) if it's an UltraSPARC as that will give you some of these instructions. Asterisk strikes me as an application that'd make fairly hefty use of things like integer multiplication. Another place to ask might be the Debian-SPARC mailing list? -- Stuart Longland (aka Redhatter, VK4MSL) .'''. Gentoo Linux/MIPS Cobalt and Docs Developer '.'` : . . . . . . . . . . . . . . . . . . . . . . .'.' http://dev.gentoo.org/~redhatter :.' I haven't lost my mind... ...it's backed up on a tape somewhere. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error compiling Asterisk 1.8.3 on Sun SPARC x64 w/Debian Squeeze
Hello Stuart On Mon, Mar 7, 2011 at 6:09 PM, Stuart Longland redhat...@gentoo.orgwrote: On 03/08/11 08:49, RR wrote: Any idea where this is coming from? seems like something is selected that doesn't have other related stuff unselected? no clue where to start looking No SPARC expert, but I seem to recall the lowest-common-denominator SPARCs lack things like hardware multiply in the instruction set. Even if it doesn't help fix the problem, you probably will want to use at least -mcpu=v9 (educated guess looking at the gcc manpage) if it's an UltraSPARC as that will give you some of these instructions. Asterisk strikes me as an application that'd make fairly hefty use of things like integer multiplication. Ok, where would I put this -mcpu=v9 in the configure line? I tried ./configure CFLAGS=-mcpu=v9? BTW, at the end of the configure script, it's already detecting the host cpu as sparc64. If that helps. Maybe -march needs to be specified somewhere? Another place to ask might be the Debian-SPARC mailing list? haha funny, I was just writing an email to that list when your email hit my inbox :) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error compiling Asterisk 1.8.3 on Sun SPARC x64 w/Debian Squeeze
On 03/08/11 09:21, RR wrote: Hello Stuart On Mon, Mar 7, 2011 at 6:09 PM, Stuart Longland redhat...@gentoo.org mailto:redhat...@gentoo.org wrote: Even if it doesn't help fix the problem, you probably will want to use at least -mcpu=v9 (educated guess looking at the gcc manpage) if it's an UltraSPARC as that will give you some of these instructions. Asterisk strikes me as an application that'd make fairly hefty use of things like integer multiplication. Ok, where would I put this -mcpu=v9 in the configure line? I tried ./configure CFLAGS=-mcpu=v9? Normally it's specified in the environment; so maybe CFLAGS=-mcpu=v9 ./configure… BTW, at the end of the configure script, it's already detecting the host cpu as sparc64. If that helps. Maybe -march needs to be specified somewhere? Maybe, the fact that it detected 'sparc64' probably is more a case of telling the build system that the system is big-endian, requires that data structures be 64-bit aligned, etc. Use of features that weren't in the first SPARC is an optional extra. Another place to ask might be the Debian-SPARC mailing list? haha funny, I was just writing an email to that list when your email hit my inbox :) Telepathy; seems we think alike. :-D Must be due to me being from the same part of the world. -- Stuart Longland (aka Redhatter, VK4MSL) .'''. Gentoo Linux/MIPS Cobalt and Docs Developer '.'` : . . . . . . . . . . . . . . . . . . . . . . .'.' http://dev.gentoo.org/~redhatter :.' I haven't lost my mind... ...it's backed up on a tape somewhere. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error compiling Asterisk 1.8.3 on Sun SPARC x64 w/Debian Squeeze
On Mon, Mar 7, 2011 at 6:31 PM, Stuart Longland redhat...@gentoo.orgwrote: On 03/08/11 09:21, RR wrote: Hello Stuart On Mon, Mar 7, 2011 at 6:09 PM, Stuart Longland redhat...@gentoo.org mailto:redhat...@gentoo.org wrote: Even if it doesn't help fix the problem, you probably will want to use at least -mcpu=v9 (educated guess looking at the gcc manpage) if it's an UltraSPARC as that will give you some of these instructions. Asterisk strikes me as an application that'd make fairly hefty use of things like integer multiplication. Ok, where would I put this -mcpu=v9 in the configure line? I tried ./configure CFLAGS=-mcpu=v9? Normally it's specified in the environment; so maybe CFLAGS=-mcpu=v9 ./configure… I tried both ways, my way and yours i.e. setting them as env variables and it still gets that error. Also found some other stuff on the net related to that in different context but none of those work for me. Some where in some old debian archives there's some mention of the Boost libraries and the flag that must be used on Sparc with Boost libraries. Although it also says that it was fixed in some later release which was back in 2008, so am assuming that fix is still in place in Squeeze. BTW, at the end of the configure script, it's already detecting the host cpu as sparc64. If that helps. Maybe -march needs to be specified somewhere? Maybe, the fact that it detected 'sparc64' probably is more a case of telling the build system that the system is big-endian, requires that data structures be 64-bit aligned, etc. Use of features that weren't in the first SPARC is an optional extra. Ok, if that doesn't help then another interesting insight is that in config.log, it says that the response to 'arch' and 'arch -k' commands is 'unknown'. Don't know if that helps. Another place to ask might be the Debian-SPARC mailing list? haha funny, I was just writing an email to that list when your email hit my inbox :) Telepathy; seems we think alike. :-D Must be due to me being from the same part of the world. Possibly :) although I have found that there's not a lot of activity in that list on a regular basis. So not sure if my problem will get resolved there or not :( -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error loading module ��Է�Vi.so
Just recently I noticed that my Asterisk 1.8 server is giving the following error at startup: [Feb 9 17:48:56] WARNING[7968]: loader.c:387 load_dynamic_module: Error loading module '��Է�Vi': /usr/lib/asterisk/modules/��Է�Vi.so: cannot open shared object file: No such file or directory I have checked the modules directory and there are no files with strange characters that I can see. The only extra modules I have are the G729 codec and Lumenvox speech recognition. Both are loaded and working. All expected functionality for Asterisk is working so I really do not know what that module may be. Any ideas? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error loading module 'cdr_radius.so'
I have this Error Please Help me loader.c: Error loading module 'cdr_radius.so': libradiusclient-ng.so.2: cannot open shared object file: No such file or directory -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error loading module 'cdr_radius.so'
Hello, you have to install radiusclient-ng http://developer.berlios.de/projects/radiusclient-ng/ Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error: Unable to create channel of type 'SIP'
Hi, I am using Trixbox 2.6.2.3, ISO install I am getting the below error in `/var/log/asterisk/full` Unable to create channel of type 'SIP' (cause 3 - No route to destination) Is there anyway to figure out which extension is causing this error ? Thank you. Best regards, Sanjay -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error: Unable to create channel of type 'SIP'
On Mon, Feb 7, 2011 at 7:42 AM, RSCL Mumbai rscl.mum...@gmail.com wrote: Hi, I am using Trixbox 2.6.2.3, ISO install I am getting the below error in `/var/log/asterisk/full` Unable to create channel of type 'SIP' (cause 3 - No route to destination) Is there anyway to figure out which extension is causing this error ? Thank you. Best regards, Sanjay -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Now,Sanjay, don't take this personally, you just happen to be ANOTHER person who has sent an email to the list lately that just crosses my tolerance for lack of respect for the try and figure it out yourself before you ask for help mentality behind this (and most Open Source project's) mailing list. ' First solution (1.5 seconds after I read your question)...check your call detail records! ,You'll see the failed call(s)! Second solution (thought of nanoseconds after the first one)NowI want you to think just a tiny bit here...If you wanted to know if a host was reachable, what would you do?.This is the same thing, except you have a list of hosts and you need to determine WHICH one cannot be reached..You try to contact each host until you find one or more that gives you a no route to host message! ping is your friend, so is mtr, also a telnet session (over the port specified for SIP to that host in your config) could be used.. Third possible method: What level of verbosity is the server currently running at? If it's not running at 3 or higher, set verbose to at least 3. That way you will see the dialplan executions that occur just before that message. Once you see that, you'll most likely have your answer. Useful tip: Another thing you could do, set qualify=yes on your sip endpoints' configurations, since this is a no route to host issue, you'll see failure on at least one of them, which will also give you your answer. Now, I'm going to sound like a jerk, but these are all simple methods that you could/should have come up with...How many seconds did you spend thinking about the issue before you decided to ask the list for help with a question that is admittedly something you should have SOME idea regarding how to test Man, I'm starting to just get pissed...That's what, 3 questions I've seen in the last 12 or less hours where the person asking the question OBVIOUSLY doesn't want to put forth any effort on their own before asking the rest of us how to do something? Asterisk Documentation is your friend! UNDERSTANDING at least 25% of how VoIP works is handy! GOOGLE is your friend! And in the name of any and all things/beings that you guys find to be holy, put forth some damned effort before asking everyone else to do the work for you Finally, if you HAVE put forth effort, LET US KNOW!!! It lessens the chance of you getting flamed by some guy who's been working for over 40 hours STRAIGHT and is just tired of seeing email after email after email containing questions that have been answered hundreds of times on the list and there are readily available answers via documentation and/or a little friggin googling.. That's it...I'm going back to barely reading the list...Every time I try to start reading it on a fairly often basis (in the hopes of being able to help people with continuing issues AFTER putting some damn effort towards the problem), I start seeing that 75-80% of new requests have 0-5% effort put forth into trying to fix it themselves, and this includes basic stuff like RTFM! Cheers -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error compiling Dahdi: invalid use of undefined type struct module
All, I'm installing Asterisk with Dahdi on a server with a custom kernel compile. I've got the kernel source in /lib/modules/2.6.34.6--grs-ipv6-64/build which points to /usr/src/linux-headers-2.6.34.6 and I think that's fine, but am getting all these struct module errors. Can anyone advise? Thanks! # make make -C drivers/dahdi/firmware firmware-loaders make[1]: entrant dans le répertoire « /usr/src/dahdi-linux-2.4.0/drivers/dahdi/firmware » make[1]: quittant le répertoire « /usr/src/dahdi-linux-2.4.0/drivers/dahdi/firmware » make -C /lib/modules/2.6.34.6--grs-ipv6-64/build SUBDIRS=/usr/src/dahdi-linux-2.4.0/drivers/dahdi DAHDI_INCLUDE=/usr/src/dahdi-linux-2.4.0/include DAHDI_MODULES_EXTRA= HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m make[1]: entrant dans le répertoire « /usr/src/linux-headers-2.6.34.6 » CC [M] /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.o /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘dahdi_register_tone_zone’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:1440: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘start_tone’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:1514: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘dahdi_chan_reg’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:1638: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘dahdi_ppp_xmit’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:1910: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:1913: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘dahdi_chan_unreg’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:2013: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘dahdi_rbs_sethook’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:2425: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:2429: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:2433: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:2477: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘dahdi_cas_setbits’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:2489: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘dahdi_timer_release’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:2732: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘dahdi_read’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:2943: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘dahdi_write’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:2974: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘ioctl_load_zone’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:3041: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:3081: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:3109: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:3137: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘dahdi_mf_tone’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:3237: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘dahdi_release’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:3460: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘dahdi_alarm_notify’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:3532: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:3544: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:3549: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:3554: error: invalid use of undefined type ‘struct module’ /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘dahdi_common_ioctl’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:4055: error: invalid use of undefined type
Re: [asterisk-users] Error compiling Dahdi: invalid use of undefined type struct module
On 1/30/11 8:45 PM, David Cunningham wrote: I'm installing Asterisk with Dahdi on a server with a custom kernel compile. I've got the kernel source in /lib/modules/2.6.34.6--grs-ipv6-64/build which points to /usr/src/linux-headers-2.6.34.6 and I think that's fine, but am getting all these struct module errors. Can anyone advise? Thanks! # make make -C drivers/dahdi/firmware firmware-loaders make[1]: entrant dans le répertoire « /usr/src/dahdi-linux-2.4.0/drivers/dahdi/firmware » make[1]: quittant le répertoire « /usr/src/dahdi-linux-2.4.0/drivers/dahdi/firmware » make -C /lib/modules/2.6.34.6--grs-ipv6-64/build SUBDIRS=/usr/src/dahdi-linux-2.4.0/drivers/dahdi DAHDI_INCLUDE=/usr/src/dahdi-linux-2.4.0/include DAHDI_MODULES_EXTRA= HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m make[1]: entrant dans le répertoire « /usr/src/linux-headers-2.6.34.6 » CC [M] /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.o /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘dahdi_register_tone_zone’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:1440: error: invalid use of undefined type ‘struct module’ Normally this is the result of not having CONFIG_MODULES set in your kernel config. This is set when you check Enable loadable module support on the top level menu in menuconfig. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error compiling Dahdi: invalid use of undefined type struct module
Shaun, CONFIG_MODULES wasn't enabled - thanks for the advice! On Mon, Jan 31, 2011 at 4:02 PM, Shaun Ruffell sruff...@digium.com wrote: On 1/30/11 8:45 PM, David Cunningham wrote: I'm installing Asterisk with Dahdi on a server with a custom kernel compile. I've got the kernel source in /lib/modules/2.6.34.6--grs-ipv6-64/build which points to /usr/src/linux-headers-2.6.34.6 and I think that's fine, but am getting all these struct module errors. Can anyone advise? Thanks! # make make -C drivers/dahdi/firmware firmware-loaders make[1]: entrant dans le répertoire « /usr/src/dahdi-linux-2.4.0/drivers/dahdi/firmware » make[1]: quittant le répertoire « /usr/src/dahdi-linux-2.4.0/drivers/dahdi/firmware » make -C /lib/modules/2.6.34.6--grs-ipv6-64/build SUBDIRS=/usr/src/dahdi-linux-2.4.0/drivers/dahdi DAHDI_INCLUDE=/usr/src/dahdi-linux-2.4.0/include DAHDI_MODULES_EXTRA= HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m make[1]: entrant dans le répertoire « /usr/src/linux-headers-2.6.34.6 » CC [M] /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.o /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c: In function ‘dahdi_register_tone_zone’: /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.c:1440: error: invalid use of undefined type ‘struct module’ Normally this is the result of not having CONFIG_MODULES set in your kernel config. This is set when you check Enable loadable module support on the top level menu in menuconfig. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error building network library on OpenSolaris and 1.8.1-rc1
Hi Tilghman, Btw w.r.t to the patch delivered for this bug, as I stated in the notes, it worked for trunk. I tried it for 1.6.2.15 and the patch came up with a few errors, as in the patch wasn't clean and I just looked at the configure.ac.rej file and made the changes manually. I wanted to test building this on Solaris 10 u9, but wasn't able to due to my messed up dev environment. I will fix this environment and test compiling and building it assuming I made the changes that the patch was supposed to make correctly. Will let you know . I was going to add that as a note to the bug report itself but then I got distracted with something else and now it's closed and I'll have to repoen it to add any more notes. Just FYI. \RR On Wed, Dec 8, 2010 at 3:55 PM, Tilghman Lesher tles...@digium.com wrote: On Wednesday 08 December 2010 14:21:57 RR wrote: Hi Guys, Any one want to take a stab at helping with this please?? All I have found so far is that the netsock.c file has code that references to taking note when it's being built on a Solaris platform, but since I don't understand this a whole lot, I am not sure where to go from here...this is the excerpt from the netsock.c file: *#if defined (SOLARIS) #include sys/sockio.h #elif defined(HAVE_GETIFADDRS) #include ifaddrs.h #endif * I would've have thought this would have taken care of the issue by making sure 'make' handles this correctly but I guess not. Anyone? Please? http://opensolaris.org/jive/thread.jspa?threadID=116059tstart=105 I suspect we'll have to make a more complex check to verify that the structure elements are all there. Please open an issue on issues.asterisk.org and reference this thread. We can then put up a patch that you can use to verify if better detection fixes your issue. Once verified, the patch will find its way into releases. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error building network library on OpenSolaris and 1.8.1-rc1
Hi RR, I've not tried compiling 1.8.1-rc1 on Solaris yet and I've not come across this issue as of yet. I did build 1.8.0-rc5 on Solaris 10 without any build error's though. I'm not sure if the code has changed that much between 1.8.0-rc5 and 1.8.1-rc1. I'm no coding guru by anyone's standards, but I do build a couple applications for Solaris. What has made my life a hell-of-a-lot easier is JDS-CBE and SFE, check out the following 2 links: http://dlc.sun.com/osol/jds/downloads/cbe/ http://pkgbuild.sourceforge.net/spec-files-extra/ What the above does is setup a common build environment for building applications. The SFE (spec-file-extra) is a framework for create rpm type spec files for solaris. Once you have one setup for asterisk then it is just a one line command to download and build asterisk. This is what I have been using to build asterisk on Solaris 10 for the past 3 years. It keeps the environment identical between versions. Have a look at getting that up and going first and then check out the spec file format and create one for your asterisk version you want to compile. My spec file is far from perfect at the moment, but it does work for what we require at the moment. Disclaimer: This is a little bit of work to setup and get working initially, but once it is setup and working, building subsequent asterisk versions and creating the Solaris SRV4 packages is a breeze :) Thanks Bruce From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RR Sent: 08 December 2010 23:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Error building network library on OpenSolaris and 1.8.1-rc1 On Wed, Dec 8, 2010 at 3:55 PM, Tilghman Lesher tles...@digium.commailto:tles...@digium.com wrote: On Wednesday 08 December 2010 14:21:57 RR wrote: Hi Guys, Any one want to take a stab at helping with this please?? All I have found so far is that the netsock.c file has code that references to taking note when it's being built on a Solaris platform, but since I don't understand this a whole lot, I am not sure where to go from here...this is the excerpt from the netsock.c file: *#if defined (SOLARIS) #include sys/sockio.h #elif defined(HAVE_GETIFADDRS) #include ifaddrs.h #endif * I would've have thought this would have taken care of the issue by making sure 'make' handles this correctly but I guess not. Anyone? Please? http://opensolaris.org/jive/thread.jspa?threadID=116059tstart=105 I suspect we'll have to make a more complex check to verify that the structure elements are all there. Please open an issue on issues.asterisk.orghttp://issues.asterisk.org/ and reference this thread. We can then put up a patch that you can use to verify if better detection fixes your issue. Once verified, the patch will find its way into releases. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.comhttp://www.digium.com/ www.asterisk.orghttp://www.asterisk.org/ G'day Tilghman, Thanks for that thread. I guess a few other things broke because of the change and the consuming application then needs to be a little smarter like you said (and suggested by darrenr) to detect whether you're on OSOL or Solaris. Does that mean I should check this same thing out on Solaris 10 as well and see what happens? I am so lost with the Solaris build environment as (and I whinged about this earlier too) there is no good way of obtaining the standard Solaris packages and dependancies and everything just goes all over the place and then one is left scurrying around to find where the damn library needs to be for it to compile. Anyway, I will open an issue and reference this thread and we'll go from there. BTW, THANK YOU for taking note of this and trying to help. You guys will have bottomless beer pitchers paid for if you guys help me get this working and are ever in the NY area :) Cheers, \R -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error building network library on OpenSolaris and 1.8.1-rc1
BTW, the issue was created yesterday, but I didn't think there was a need to post it here but nevertheless for posterity, the Issue ID is: 18442 Thanks \RR On Wed, Dec 8, 2010 at 6:57 PM, RR ranjt...@gmail.com wrote: On Wed, Dec 8, 2010 at 3:55 PM, Tilghman Lesher tles...@digium.comwrote: On Wednesday 08 December 2010 14:21:57 RR wrote: Hi Guys, Any one want to take a stab at helping with this please?? All I have found so far is that the netsock.c file has code that references to taking note when it's being built on a Solaris platform, but since I don't understand this a whole lot, I am not sure where to go from here...this is the excerpt from the netsock.c file: *#if defined (SOLARIS) #include sys/sockio.h #elif defined(HAVE_GETIFADDRS) #include ifaddrs.h #endif * I would've have thought this would have taken care of the issue by making sure 'make' handles this correctly but I guess not. Anyone? Please? http://opensolaris.org/jive/thread.jspa?threadID=116059tstart=105 I suspect we'll have to make a more complex check to verify that the structure elements are all there. Please open an issue on issues.asterisk.org and reference this thread. We can then put up a patch that you can use to verify if better detection fixes your issue. Once verified, the patch will find its way into releases. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org G'day Tilghman, Thanks for that thread. I guess a few other things broke because of the change and the consuming application then needs to be a little smarter like you said (and suggested by darrenr) to detect whether you're on OSOL or Solaris. Does that mean I should check this same thing out on Solaris 10 as well and see what happens? I am so lost with the Solaris build environment as (and I whinged about this earlier too) there is no good way of obtaining the standard Solaris packages and dependancies and everything just goes all over the place and then one is left scurrying around to find where the damn library needs to be for it to compile. Anyway, I will open an issue and reference this thread and we'll go from there. BTW, THANK YOU for taking note of this and trying to help. You guys will have bottomless beer pitchers paid for if you guys help me get this working and are ever in the NY area :) Cheers, \R -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error building network library on OpenSolaris and 1.8.1-rc1
On Thu, Dec 9, 2010 at 10:02 AM, Bruce McAlister bruce.mcalis...@blueface.ie wrote: Hi RR, I’ve not tried compiling 1.8.1-rc1 on Solaris yet and I’ve not come across this issue as of yet. I did build 1.8.0-rc5 on Solaris 10 without any build error’s though. I’m not sure if the code has changed that much between 1.8.0-rc5 and 1.8.1-rc1. I’m no coding guru by anyone’s standards, but I do build a couple applications for Solaris. What has made my life a hell-of-a-lot easier is JDS-CBE and SFE, check out the following 2 links: http://dlc.sun.com/osol/jds/downloads/cbe/ http://pkgbuild.sourceforge.net/spec-files-extra/ What the above does is setup a common build environment for building applications. The SFE (spec-file-extra) is a framework for create rpm type spec files for solaris. Once you have one setup for asterisk then it is just a one line command to download and build asterisk. This is what I have been using to build asterisk on Solaris 10 for the past 3 years. It keeps the environment identical between versions. Have a look at getting that up and going first and then check out the spec file format and create one for your asterisk version you want to compile. My spec file is far from perfect at the moment, but it does work for what we require at the moment. Disclaimer: This is a little bit of work to setup and get working initially, but once it is setup and working, building subsequent asterisk versions and creating the Solaris SRV4 packages is a breeze J Thanks Bruce Hi Bruce, Thanks so much for that. I don't know what to tell you as to why I'm getting the error if you didn't. Maybe it's because I'm using OpenSolaris as opposed to Solaris? That's the only thing I can think of and Tilghman's comment also kind of hinted at that the Makefile and/or configure or the overall build process needs to be smarter to tell when the system is being built for Solaris or OpenSolaris. Also while searching for something else but a related issue, I found another thread that had talked about successfully compiling 1.8 beta on Solaris on Sparc. So there's definitely hope. But I think this might be an OpenSolaris thing as even though I don't have the sophistication of CBE and Sun Studio etc, I do have the reasonably convenient VM snapshots to get a clean system whenever I want to and I can tell you, there was NOTHING on this system other than a fresh OpenSolaris install, and the gcc-dev package. Hmm Anyway, let's see if the nice developers at Digium can find some time to put in a fix for this so the product might become buildable over Solaris AND OpenSolaris and people can then just go with the platform of their choice. Cheers, RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error building network library on OpenSolaris and 1.8.1-rc1
Hello All, I have been banging my head against trying to get asterisk to compile on Solaris as well as OpenSolaris. I've tried to build various versions of Asterisk as on various versions of Solaris and OpenSolaris to no avail. Finally, I said, what the heck, I got the latest version of OpenSolaris that (pkg image-update) could get and then the latest ver of asterisk I found on the digium repo. Amazingly, configure and make menuselect went without a hitch, very clean. 'make' was going really well as well, in fact this is the farthest I've ever seen it ever go with the minor hitch compalining about format_mp3 but it suggested I use that script in contrib and download the code for that and that made it run again. BUT just my luck, it crapped out with this error *netsock.c: In function `ast_set_default_eid': netsock.c:250: error: structure has no member named `ifr_hwaddr' make[1]: *** [netsock.o] Error 1 make: *** [main] Error 2 * Can anyone please help me resolve this? I don't even know where to look. Google came back with nothing. Same with a search through the 30,000+ emails I have from the Asterisk mailing list only gave me the hint that it's a function from if.h which in OpenSolaris resides in /usr/include/net as opposed to maybe /usr/include/linux. Any ideas? Thanks RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error building network library on OpenSolaris and 1.8.1-rc1
Hi Guys, Any one want to take a stab at helping with this please?? All I have found so far is that the netsock.c file has code that references to taking note when it's being built on a Solaris platform, but since I don't understand this a whole lot, I am not sure where to go from here...this is the excerpt from the netsock.c file: *#if defined (SOLARIS) #include sys/sockio.h #elif defined(HAVE_GETIFADDRS) #include ifaddrs.h #endif * I would've have thought this would have taken care of the issue by making sure 'make' handles this correctly but I guess not. Anyone? Please? Thanks \RR On Wed, Dec 8, 2010 at 4:43 AM, RR ranjt...@gmail.com wrote: Hello All, I have been banging my head against trying to get asterisk to compile on Solaris as well as OpenSolaris. I've tried to build various versions of Asterisk as on various versions of Solaris and OpenSolaris to no avail. Finally, I said, what the heck, I got the latest version of OpenSolaris that (pkg image-update) could get and then the latest ver of asterisk I found on the digium repo. Amazingly, configure and make menuselect went without a hitch, very clean. 'make' was going really well as well, in fact this is the farthest I've ever seen it ever go with the minor hitch compalining about format_mp3 but it suggested I use that script in contrib and download the code for that and that made it run again. BUT just my luck, it crapped out with this error *netsock.c: In function `ast_set_default_eid': netsock.c:250: error: structure has no member named `ifr_hwaddr' make[1]: *** [netsock.o] Error 1 make: *** [main] Error 2 * Can anyone please help me resolve this? I don't even know where to look. Google came back with nothing. Same with a search through the 30,000+ emails I have from the Asterisk mailing list only gave me the hint that it's a function from if.h which in OpenSolaris resides in /usr/include/net as opposed to maybe /usr/include/linux. Any ideas? Thanks RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error building network library on OpenSolaris and 1.8.1-rc1
On 10-12-08 03:21 PM, RR wrote: Any one want to take a stab at helping with this please?? All I have found so far is that the netsock.c file has code that references to taking note when it's being built on a Solaris platform, but since I don't understand this a whole lot, I am not sure where to go from here...this is the excerpt from the netsock.c file: I'm in the process of bring up our remote Bamboo agents for Solaris, so I can see if I get the same issue. Which versions of Solaris and OpenSolaris are you using? -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error building network library on OpenSolaris and 1.8.1-rc1
On Wed, Dec 8, 2010 at 3:40 PM, Paul Belanger pabelan...@digium.com wrote: On 10-12-08 03:21 PM, RR wrote: Any one want to take a stab at helping with this please?? All I have found so far is that the netsock.c file has code that references to taking note when it's being built on a Solaris platform, but since I don't understand this a whole lot, I am not sure where to go from here...this is the excerpt from the netsock.c file: I'm in the process of bring up our remote Bamboo agents for Solaris, so I can see if I get the same issue. Which versions of Solaris and OpenSolaris are you using? -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org Hi Paul, I haven't tried compiling it on Solaris 10 as yet, as OpenSolaris is a lot easier to update and download packages / dependencies etc. neverthess, the OpenSolaris version is: OpenSolaris 2010.05 snv_134b X86, running on a Core 2 Duo Quad machine inside a 64-bit Hyper-V VM. Let me know if you need more info. BTW, the way the OS was installed was through the ISO available on the OpenSolaris website and then updating it with 'pkg image-update' command and then following it with installing the gcc-dev package. Thanks \RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error building network library on OpenSolaris and 1.8.1-rc1
On Wed, Dec 8, 2010 at 3:55 PM, Tilghman Lesher tles...@digium.com wrote: On Wednesday 08 December 2010 14:21:57 RR wrote: Hi Guys, Any one want to take a stab at helping with this please?? All I have found so far is that the netsock.c file has code that references to taking note when it's being built on a Solaris platform, but since I don't understand this a whole lot, I am not sure where to go from here...this is the excerpt from the netsock.c file: *#if defined (SOLARIS) #include sys/sockio.h #elif defined(HAVE_GETIFADDRS) #include ifaddrs.h #endif * I would've have thought this would have taken care of the issue by making sure 'make' handles this correctly but I guess not. Anyone? Please? http://opensolaris.org/jive/thread.jspa?threadID=116059tstart=105 I suspect we'll have to make a more complex check to verify that the structure elements are all there. Please open an issue on issues.asterisk.org and reference this thread. We can then put up a patch that you can use to verify if better detection fixes your issue. Once verified, the patch will find its way into releases. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org G'day Tilghman, Thanks for that thread. I guess a few other things broke because of the change and the consuming application then needs to be a little smarter like you said (and suggested by darrenr) to detect whether you're on OSOL or Solaris. Does that mean I should check this same thing out on Solaris 10 as well and see what happens? I am so lost with the Solaris build environment as (and I whinged about this earlier too) there is no good way of obtaining the standard Solaris packages and dependancies and everything just goes all over the place and then one is left scurrying around to find where the damn library needs to be for it to compile. Anyway, I will open an issue and reference this thread and we'll go from there. BTW, THANK YOU for taking note of this and trying to help. You guys will have bottomless beer pitchers paid for if you guys help me get this working and are ever in the NY area :) Cheers, \R -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error messages with chan_dahdi
On Sat, Dec 4, 2010 at 2:11 PM, Shaun Ruffell sruff...@digium.com wrote: On 12/4/10 9:15 AM, equis software wrote: HI, I'm using asterisk-1.4.24, dahdi-linux-complete-2.4.0+2.4.0 and libpri-1.4.11.4 When dial, when 492131 answer, in console appear some error messages -- AGI Script Executing Application: (DIAL) Options: (DAHDI/g1/492131|60) -- Requested transfer capability: 0x00 - SPEECH -- Called g1/492131 [Dec 4 11:15:59] WARNING[7669]: chan_dahdi.c:1776 dahdi_enable_ec: Unable to enable echo cancellation on channel 2 (No such device) -- DAHDI/2-1 is ringing [Dec 4 11:16:02] WARNING[7669]: chan_dahdi.c:1776 dahdi_enable_ec: Unable to enable echo cancellation on channel 2 (No such device) These No such device errors when trying to enable the echocan are most likely the result of not having configured an echocan for the channel in /etc/dahdi/system.conf. See line 309 in http://svn.asterisk.org/view/dahdi/tools/trunk/system.conf.sample?view=markup Yes, I wasn't cofigure the echocan, thanks! -- DAHDI/2-1 answered DAHDI/1-1 -- Native bridging DAHDI/1-1 and DAHDI/2-1 [Dec 4 11:16:02] ERROR[7669]: chan_dahdi.c:8735 dahdi_pri_error: ROSE REJECT: [Dec 4 11:16:02] ERROR[7669]: chan_dahdi.c:8735 dahdi_pri_error: INVOKE ID: 3 [Dec 4 11:16:02] ERROR[7669]: chan_dahdi.c:8735 dahdi_pri_error: PROBLEM: Invoke: Unrecognized Operation These errors I don't know about off the top of my head (and they are probably the more critical ones for you I'm guessing). Yes again, this is my real problem, despite the error messages,I can make the call without problems, but I'm worried about it... -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error loading module 'chan_gtalk.so': libiksemel.so.3: cannot open shared object file: No such file or directory
Yes sir, We are pass the error. Works like a charm. I just documented this on our new wiki: http://voipcomsolutions.com/wiki/index.php?title=How_to_install_Asterisk_from_source_-_Google_Integration_ready Thanks again *José Pablo Méndez * 2010/12/1 José Pablo Méndez Soto aux...@gmail.com Thank you sir, I got to read your email a few minutes ago. I will try your recommendation and update. On Tue, Nov 30, 2010 at 5:01 PM, Tilghman Lesher tles...@digium.comwrote: On Tuesday 30 November 2010 16:27:33 José Pablo Méndez Soto wrote: Sorry never mind! I got it to work after sof-linking to /lib/, and loading res_jabber.so first, chan_gtalk.so second. So in summary: ln -s /usr/local/lib /lib/ The better way to do this would be: echo /usr/local/lib /etc/ld.so.conf.d/iksemel.conf ldconfig -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error messages with chan_dahdi
On 12/4/10 9:15 AM, equis software wrote: HI, I'm using asterisk-1.4.24, dahdi-linux-complete-2.4.0+2.4.0 and libpri-1.4.11.4 When dial, when 492131 answer, in console appear some error messages -- AGI Script Executing Application: (DIAL) Options: (DAHDI/g1/492131|60) -- Requested transfer capability: 0x00 - SPEECH -- Called g1/492131 [Dec 4 11:15:59] WARNING[7669]: chan_dahdi.c:1776 dahdi_enable_ec: Unable to enable echo cancellation on channel 2 (No such device) -- DAHDI/2-1 is ringing [Dec 4 11:16:02] WARNING[7669]: chan_dahdi.c:1776 dahdi_enable_ec: Unable to enable echo cancellation on channel 2 (No such device) These No such device errors when trying to enable the echocan are most likely the result of not having configured an echocan for the channel in /etc/dahdi/system.conf. See line 309 in http://svn.asterisk.org/view/dahdi/tools/trunk/system.conf.sample?view=markup -- DAHDI/2-1 answered DAHDI/1-1 -- Native bridging DAHDI/1-1 and DAHDI/2-1 [Dec 4 11:16:02] ERROR[7669]: chan_dahdi.c:8735 dahdi_pri_error: ROSE REJECT: [Dec 4 11:16:02] ERROR[7669]: chan_dahdi.c:8735 dahdi_pri_error: INVOKE ID: 3 [Dec 4 11:16:02] ERROR[7669]: chan_dahdi.c:8735 dahdi_pri_error: PROBLEM: Invoke: Unrecognized Operation These errors I don't know about off the top of my head (and they are probably the more critical ones for you I'm guessing). -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error loading module 'chan_gtalk.so': libiksemel.so.3: cannot open shared object file: No such file or directory
Hello, Can't get chan_gtalk.so module to load, neither res_jabber.so: Asterisk*CLI module load chan_gtalk.so Unable to load module chan_gtalk.so Command 'module load chan_gtalk.so ' failed. [Dec 1 16:10:05] WARNING[2931]: loader.c:387 load_dynamic_module: Error loading module 'chan_gtalk.so': libiksemel.so.3: cannot open shared object file: No such file or directory [Dec 1 16:10:05] WARNING[2931]: loader.c:839 load_resource: Module 'chan_gtalk.so' could not be loaded. I got pass the module compilation after installing iksemel from tar ( http://code.google.com/p/iksemel/). Menuselect showed chan_gtalk check-able instead of XXX, which is good AFAIK. Also, Asterisk recognizes the modules just fine: Asterisk*CLI module load res_ res_adsi.sores_ael_share.so res_agi.so res_clialiases.so res_clioriginate.sores_convert.so res_crypto.so res_fax.so res_jabber.so res_limit.so res_monitor.so res_musiconhold.so res_mutestream.so res_phoneprov.so res_realtime.so res_rtp_asterisk.sores_rtp_multicast.so res_security_log.so res_smdi.sores_speech.so res_stun_monitor.so res_timing_dahdi.sores_timing_pthread.so res_timing_timerfd.so res_calendar.so Asterisk*CLI module load ch chan_agent.so chan_bridge.so chan_gtalk.so chan_iax2.so chan_jingle.so chan_local.so chan_mgcp.so chan_multicast_rtp.so chan_oss.so chan_phone.so chan_sip.sochan_skinny.so chan_unistim.sochan_dahdi.so Also, I made sure SSL libraries are in place: r...@asterisk:/etc/asterisk# dpkg -l openssl* libssl* ||/ NameVersion Description +++-===-===-== unlibssl none (no description available) ii libssl-dev 0.9.8g-16ubuntu3.4 SSL development libraries, header files and documentation ii libssl0.9.8 0.9.8g-16ubuntu3.4 SSL shared libraries unlibssl08-devnone (no description available) unlibssl09-devnone (no description available) unlibssl095a-dev none (no description available) unlibssl096-dev none (no description available) ii openssl 0.9.8g-16ubuntu3.4 Secure Socket Layer (SSL) binary and related cryptographic tools un openssl-doc none (no description available) iksemel was successfully installed: r...@asterisk:/etc/asterisk# ls /usr/local/lib/ libiksemel.a libiksemel.la libiksemel.so libiksemel.so.3 libiksemel.so.3.1.1 pkgconfig python2.6 Should I soft-link this libraries at another directory for Asterisk to find them? I found where chan_gtalk.so module gets the libraries from: r...@asterisk:/usr/lib/asterisk/modules# ldd chan_gtalk.so ldd chan_gtalk.so linux-vdso.so.1 = (0x7fff61bff000) libiksemel.so.3 = (Not found) libssl.so.0.9.8 = /lib/libssl.so.0.9.8 (0x7f7fd4ee6000) libcrypto.so.0.9.8 = /lib/libcrypto.so.0.9.8 (0x7f7fd4b5e000) libpthread.so.0 = /lib/libpthread.so.0 (0x7f7fd4942000) libc.so.6 = /lib/libc.so.6 (0x7f7fd45d2000) libdl.so.2 = /lib/libdl.so.2 (0x7f7fd43cd000) libz.so.1 = /lib/libz.so.1 (0x7f7fd41b6000) /lib64/ld-linux-x86-64.so.2 (0x7f7fd5559000) So I soft-linked under /lib/, and get a different error when loading the module: Asterisk*CLI module load chan_gtalk.so Unable to load module chan_gtalk.so Command 'module load chan_gtalk.so ' failed. [Dec 1 16:28:26] WARNING[3055]: loader.c:449 load_dynamic_module: Error loading module 'chan_gtalk.so': /usr/lib/asterisk/modules/chan_gtalk.so: undefined symbol: ast_aji_get_client [Dec 1 16:28:26] WARNING[3055]: loader.c:839 load_resource: Module 'chan_gtalk.so' could not be loaded. r...@asterisk:/usr/lib/asterisk/modules# !ldd ldd chan_gtalk.so linux-vdso.so.1 = (0x7fff61bff000) libiksemel.so.3 = /lib/libiksemel.so.3 (0x7f7fd5135000) --- It finds the library allright! libssl.so.0.9.8 = /lib/libssl.so.0.9.8 (0x7f7fd4ee6000) libcrypto.so.0.9.8 = /lib/libcrypto.so.0.9.8 (0x7f7fd4b5e000) libpthread.so.0 = /lib/libpthread.so.0 (0x7f7fd4942000) libc.so.6 = /lib/libc.so.6 (0x7f7fd45d2000) libdl.so.2 = /lib/libdl.so.2 (0x7f7fd43cd000) libz.so.1 = /lib/libz.so.1 (0x7f7fd41b6000) /lib64/ld-linux-x86-64.so.2 (0x7f7fd5559000) Any thoughts? *José Pablo Méndez * -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com
[asterisk-users] Error loading module 'chan_gtalk.so': libiksemel.so.3: cannot open shared object file: No such file or directory
Sorry never mind! I got it to work after sof-linking to /lib/, and loading res_jabber.so first, chan_gtalk.so second. So in summary: ln -s /usr/local/lib /lib/ asterisk-climodules load res_jabber.so asterisk-climodules load chan_gtalk.so Cheers! *José Pablo Méndez * -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error with Connecting Two Asterisk BOX with IAX
Hi bakko, just as a test, try to add calltokenoptional=0.0.0.0/0.0.0.0 to your iax.conf. Giorgio Incantalupo bakko wrote: Hello, I'm trying to conect two 1.6.2.13 Asterisk server with IAX. This is my configuration: Asterisk A: iax.conf register = coiax:pa...@69.164.207.166 [smiax] type=friend host=dynamic trunk=yes secret=pass2 context=phones deny=0.0.0.0/0.0.0.0 permit=69.164.207.166/255.255.255.255 qualify=yes Console: iax2 registry 69.164.207.166:4569 N coiax 69.164.197.105:456960 Registered iax2 peers smiax69.164.207.166 (D) 255.255.255.255 4569 (T) OK (3 ms) Asterisk B: register = smiax:pa...@69.164.197.105 [coiax] type=friend host=dynamic trunk=yes secret=pass1 context=phones deny=0.0.0.0/0.0.0.0 permit=69.164.197.105/255.255.255.255 qualify=yes Console iax2 registry 69.164.197.105:4569 N smiax 69.164.207.166:456960 Registered iax2 peers coiax69.164.197.105 (D) 255.255.255.255 4569 (T) OK (3 ms) When I try to call from Asterisk A to Asterisk B I receive this error Asterisk A WARNING[28759]: chan_iax2.c:10287 socket_process: Call rejected by 69.164.207.166: No authority found AsteriskB NOTICE[26563]: chan_iax2.c:10522 socket_process: Host 69.164.197.105 failed to authenticate as coiax What's wrong? Thank you in advance. Regards - Bakko -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error with Connecting Two Asterisk BOX with IAX
Yhank you very much Giorgio, now work with the general option: calltokenoptional=0.0.0.0/0.0.0.0 Regards - Bakko -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error with Connecting Two Asterisk BOX with IAX
Hello, I'm trying to conect two 1.6.2.13 Asterisk server with IAX. This is my configuration: Asterisk A: iax.conf register = coiax:pa...@69.164.207.166 [smiax] type=friend host=dynamic trunk=yes secret=pass2 context=phones deny=0.0.0.0/0.0.0.0 permit=69.164.207.166/255.255.255.255 qualify=yes Console: iax2 registry 69.164.207.166:4569 N coiax 69.164.197.105:456960 Registered iax2 peers smiax69.164.207.166 (D) 255.255.255.255 4569 (T) OK (3 ms) Asterisk B: register = smiax:pa...@69.164.197.105 [coiax] type=friend host=dynamic trunk=yes secret=pass1 context=phones deny=0.0.0.0/0.0.0.0 permit=69.164.197.105/255.255.255.255 qualify=yes Console iax2 registry 69.164.197.105:4569 N smiax 69.164.207.166:456960 Registered iax2 peers coiax69.164.197.105 (D) 255.255.255.255 4569 (T) OK (3 ms) When I try to call from Asterisk A to Asterisk B I receive this error Asterisk A WARNING[28759]: chan_iax2.c:10287 socket_process: Call rejected by 69.164.207.166: No authority found AsteriskB NOTICE[26563]: chan_iax2.c:10522 socket_process: Host 69.164.197.105 failed to authenticate as coiax What's wrong? Thank you in advance. Regards - Bakko -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error loading skype_for_asterisk
This suddenly started appearing and I'm not sure why. Any ideas? asterisk*CLI module load chan_skype.so Unable to load module chan_skype.so Command 'module load chan_skype.so' failed. [Sep 15 11:08:25] WARNING[12274]: loader.c:429 load_dynamic_module: Error loading module 'chan_skype.so': /usr/lib/asterisk/modules/chan_skype.so: undefined symbol: sfa_send_chat_message [Sep 15 11:08:25] WARNING[12274]: loader.c:797 load_resource: Module 'chan_skype.so' could not be loaded. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error loading skype_for_asterisk
On 09/15/2010 10:09 AM, Richard Kenner wrote: This suddenly started appearing and I'm not sure why. Any ideas? asterisk*CLI module load chan_skype.so Unable to load module chan_skype.so Command 'module load chan_skype.so' failed. [Sep 15 11:08:25] WARNING[12274]: loader.c:429 load_dynamic_module: Error loading module 'chan_skype.so': /usr/lib/asterisk/modules/chan_skype.so: undefined symbol: sfa_send_chat_message [Sep 15 11:08:25] WARNING[12274]: loader.c:797 load_resource: Module 'chan_skype.so' could not be loaded. You don't have a matching version of res_skypeforasterisk loaded. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error - Failed to extend from xxx to xxx
Hi Tilghman, Firstly, thank you very much for your answer. I had assumed it was a memory buffer but then moved my attention to the database / database connectivity. (The two might still be connected.) Would it be correct to say that adjusting some of the memory allocation settings for the kernel (in sysctl.conf) will rectify this situation? Do you have any pointers which are more specific as to which buffer it is likely to be? 1.6.1.6 is pretty old, and you should, at a minimum, upgrade to the latest 1.6.1 release, which is 1.6.1.20. There are a myriad of bugs that have been corrected in that time. That said, the 1.6.1 branch is in security mode. Even if you found a legitimate bug, it is not going to be fixed in branch, and there will be no more releases of the 1.6.1 branch, other than for security fixes. Unfortunately the machine is in production and the version can't be changed. Other than this buffer issue which only came to light recently we have managed to work around any of the other bugs that affected our installation (including a MWI bug). Changing Asterisk version would probably require an additional 100+ hours of testing to make sure everything still works properly. Again, thanks for your response. Kind Regards Stuart Elvish -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error - Failed to extend from xxx to xxx
On Sunday 27 June 2010 06:54:32 Stuart Elvish wrote: Hi Tilghman, Firstly, thank you very much for your answer. I had assumed it was a memory buffer but then moved my attention to the database / database connectivity. (The two might still be connected.) Would it be correct to say that adjusting some of the memory allocation settings for the kernel (in sysctl.conf) will rectify this situation? Not having a notion of the actual issue, I have no idea what will solve it. I can only point you to ast_str_helper() as being the cause of the message, which is the root function underlying the ast_str_set() and ast_str_append functions, which implement dynamic strings in Asterisk. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error - Failed to extend from xxx to xxx
Hi Tilghman Not having a notion of the actual issue, I have no idea what will solve it. The notification occurs when the system is idle (ie taking and making no calls) so the only logical conclusion is that it is handling registration data when the notification is printed. Also, given that a different number of error messages appear at different times, it would appear that it relates to different blocks of equipment re-registering. My only concern is if it turns into something more serious than just warning messages and stops the system from working. At the moment the client is adding more extensions so the system load is increasing incrementally hence why the problem wasn't detected initially in our testing. I can only point you to ast_str_helper() as being the cause of the message, which is the root function underlying the ast_str_set() and ast_str_append functions, which implement dynamic strings in Asterisk. I had a look at the code in ast_str_helper (saw it before posting because it matches the failed to extend search phrase) and seeing as the buffer is dynamic, wouldn't it mean that the operating system / kernel is restricting the buffer from expanding? Or is this configurable in Asterisk at compile time, that is to say that Asterisk sets the buffer size when it is compiled on the target machine based on the current kernel / operating system settings? Or, does Asterisk check this each time it reloads so it wouldn't be necessary to recompile if we found a way to increase the buffer? Thanks. Stuart Elvish -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error - Failed to extend from xxx to xxx
Hi List, I have a problem with Asterisk 1.6.1.6 realtime (MySQL databases hosted on a separate machine). When Asterisk is in verbose mode, it prints messages saying failed to extend from 512 to 664 (quite a few lines in a block) and then the last message is mostly failed to extend from 512 to 663. The number of lines varies unpredictably. The full message (in the logs) is: [Jun 26 04:02:30] VERBOSE[3257] utils.c: failed to extend from 512 to 664 I haven't been able to track down much information about this using web searches but it appears (based on what I have read) that perhaps it is a database connection issue. There doesn't appear to be any problem in terms of network connectivity or database load on the database server which would lead to this situation. The Asterisk server is also well resourced generally running at 10% load. The system is designed for high availability so the extensions re-register quite frequently (which is no problem as the extensions are on an internal network) and there are approximately 1,570 extensions with more being added each week. We use cached RT peers. The database is used for CDR's, sip.conf and voicemail.conf but extensions.conf is static. So with all the above information, I am leaning towards the error being related to the database connection for real time and it occurring when an extension re-registers. Any thoughts? Thanks in advance. Stuart Elvish -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error - Failed to extend from xxx to xxx
On Saturday 26 June 2010 04:28:18 Stuart Elvish wrote: I have a problem with Asterisk 1.6.1.6 realtime (MySQL databases hosted on a separate machine). When Asterisk is in verbose mode, it prints messages saying failed to extend from 512 to 664 (quite a few lines in a block) and then the last message is mostly failed to extend from 512 to 663. The number of lines varies unpredictably. 1.6.1.6 is pretty old, and you should, at a minimum, upgrade to the latest 1.6.1 release, which is 1.6.1.20. There are a myriad of bugs that have been corrected in that time. That said, the 1.6.1 branch is in security mode. Even if you found a legitimate bug, it is not going to be fixed in branch, and there will be no more releases of the 1.6.1 branch, other than for security fixes. The exact error that you're looking at is a memory management issue. A dynamic buffer, which started as size 512, needed to be expanded, but the memory allocation failed in some way, and this warning message was the result. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error trying to add context: Context 'internal' tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri|*|*'
Hello again dear list.
Could you please help with this?
Thank you for all support, you are great, and i am now at a late stage in the
setup and tweaking this server,
So I hope you can help me again.
I Can't make include the context nighttime. Just to demonstrate if it works, I
have a playback function there.
But CLI reports:
CLI
[Jun 18 14:20:22] WARNING[2287]: pbx.c:9542 ast_context_verify_includes:
Context 'internal' tries to include nonexistent context
'nighttime|12:30-8:00|mon-fri|*|*'
Extensions.conf
[general]
autofallthrough=yes
[default]
[incoming_calls]
[phones]
include = internal
include = hovedmeny
[internal]
include = to_SIPtrunk
include = nighttime
exten = _10X,1,NoOp()
exten = _10X,n,Dial(SIP/${EXTEN},10)
exten = _10X,n,Playback(kuntiltestt_)
;exten = _10X,n,Playback(vm-nobodyavailtt-monkeysintrott-monkeys)
exten = _10X,n,Hangup()
exten = 4767209600,1,NoOp();
exten = 4767209600,n,Verbose(Callerid num ${CALLERID(num)});
exten = 4767209600,n,Dial(SIP/501,5);
;exten = 4767209600,n,Background(velkommen_abacustast123vent_);
;exten = 4767209600,n,WaitExten;
;exten = 4767209600,1,Dial(SIP/200,15);
;exten = 4767209600,1,Goto(submenu,s,1);
exten = 4767209600,n,Playback(kuntiltestt_);
exten = 4767209600,n,Hangup();
[hovedmeny]
exten = 501,1,Answer
exten = 501,n,Wait(2)
exten = 501,n,Playback(velkommen_abacus)
exten = 501,n,Set(Loop=0)
exten = 501,n,While($[${Loop} 3])
exten = 501,n,Background(tast123vent_)
exten = 501,n,WaitExten(5)
exten = 501,n,Set(Loop=$[${Loop}+1])
exten = 501,n(LoopEnd),EndWhile()
exten = 501,n,Hangup()
exten = 1,1,Playback(tt-weasels)
exten = 1,2,Dial(SIP/200,10,rg)
exten = 1,3,Hangup()
exten = 2,1,Playback(tt-monkeys)
exten = 2,n,Dial(SIP/302,60,rg)
exten = 2,n,Hangup()
exten = 3,1,Dial(SIP/402,60,rg)
exten = 3,n,Hangup
exten = 9,n,Hangup()
exten = i,1,Set(Loop=$[${Loop}+1])
exten = i,n,Goto(LoopEnd)
exten = t,1,Set(Loop=$[${Loop}+1])
exten = t,n,Goto(LoopEnd)
[nighttime]
exten = s,1,Wait(2);
exten = s,n,Playback(tt-somethingwrong);
exten = s,n,Hangup;
[to_SIPtrunk]
exten = _[2-9]XXX, 1, Macro(dial-trunk-sip,${EXTEN});
exten = _0, 1, Macro(dial-trunk-sip,${EXTEN});
[incoming]
exten = s,1,Noop();
exten = s,n,Verbose(Call ${EXTEN});
exten = s,n,Dial(SIP/501);
exten = s,n,Hangup();
[macro-dial-trunk-sip]
exten = s,1,Noop(${ARG1},${CALLERID(num)})
exten = s,n,Set(CALLERID(num)=67209600)
exten = s,n,Dial(SIP/phonect_01/${ARG1})
exten = s,n,Hangup
exten = s,n,MacroExit
Med vennlig hilsen / Best regards
Abacus IT AS
- din Visma Software Partner
- your Visma Software Partner
Tor Aksel Celasun
Mobilnummer/cell phone: (+47) 900 15 103
Sentralbord/Support 4000 1850
ak...@abacus-it.nomailto:ak...@abacus-it.no
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Re: [asterisk-users] Error trying to add context: Context 'internal' tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri|*|*'
Minor edit on the include = nighttime|12:30-8:00|mon-fri|*|*
Correct now.
Fra: Aksel Celasun
Sendt: 18. juni 2010 14:30
Til: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Emne: Error trying to add context: Context 'internal' tries to include
nonexistent context 'nighttime|12:30-8:00|mon-fri|*|*'
Hello again dear list.
Could you please help with this?
Thank you for all support, you are great, and i am now at a late stage in the
setup and tweaking this server,
So I hope you can help me again.
I Can't make include the context nighttime. Just to demonstrate if it works, I
have a playback function there.
But CLI reports:
CLI
[Jun 18 14:20:22] WARNING[2287]: pbx.c:9542 ast_context_verify_includes:
Context 'internal' tries to include nonexistent context
'nighttime|12:30-8:00|mon-fri|*|*'
Extensions.conf
[general]
autofallthrough=yes
[default]
[incoming_calls]
[phones]
include = internal
include = hovedmeny
[internal]
include = to_SIPtrunk
include = nighttime|12:30-8:00|mon-fri|*|*
exten = _10X,1,NoOp()
exten = _10X,n,Dial(SIP/${EXTEN},10)
exten = _10X,n,Playback(kuntiltestt_)
;exten = _10X,n,Playback(vm-nobodyavailtt-monkeysintrott-monkeys)
exten = _10X,n,Hangup()
exten = 4767209600,1,NoOp();
exten = 4767209600,n,Verbose(Callerid num ${CALLERID(num)});
exten = 4767209600,n,Dial(SIP/501,5);
;exten = 4767209600,n,Background(velkommen_abacustast123vent_);
;exten = 4767209600,n,WaitExten;
;exten = 4767209600,1,Dial(SIP/200,15);
;exten = 4767209600,1,Goto(submenu,s,1);
exten = 4767209600,n,Playback(kuntiltestt_);
exten = 4767209600,n,Hangup();
[hovedmeny]
exten = 501,1,Answer
exten = 501,n,Wait(2)
exten = 501,n,Playback(velkommen_abacus)
exten = 501,n,Set(Loop=0)
exten = 501,n,While($[${Loop} 3])
exten = 501,n,Background(tast123vent_)
exten = 501,n,WaitExten(5)
exten = 501,n,Set(Loop=$[${Loop}+1])
exten = 501,n(LoopEnd),EndWhile()
exten = 501,n,Hangup()
exten = 1,1,Playback(tt-weasels)
exten = 1,2,Dial(SIP/200,10,rg)
exten = 1,3,Hangup()
exten = 2,1,Playback(tt-monkeys)
exten = 2,n,Dial(SIP/302,60,rg)
exten = 2,n,Hangup()
exten = 3,1,Dial(SIP/402,60,rg)
exten = 3,n,Hangup
exten = 9,n,Hangup()
exten = i,1,Set(Loop=$[${Loop}+1])
exten = i,n,Goto(LoopEnd)
exten = t,1,Set(Loop=$[${Loop}+1])
exten = t,n,Goto(LoopEnd)
[nighttime]
exten = s,1,Wait(2);
exten = s,n,Playback(tt-somethingwrong);
exten = s,n,Hangup;
[to_SIPtrunk]
exten = _[2-9]XXX, 1, Macro(dial-trunk-sip,${EXTEN});
exten = _0, 1, Macro(dial-trunk-sip,${EXTEN});
[incoming]
exten = s,1,Noop();
exten = s,n,Verbose(Call ${EXTEN});
exten = s,n,Dial(SIP/501);
exten = s,n,Hangup();
[macro-dial-trunk-sip]
exten = s,1,Noop(${ARG1},${CALLERID(num)})
exten = s,n,Set(CALLERID(num)=67209600)
exten = s,n,Dial(SIP/phonect_01/${ARG1})
exten = s,n,Hangup
exten = s,n,MacroExit
Med vennlig hilsen / Best regards
Abacus IT AS
- din Visma Software Partner
- your Visma Software Partner
Tor Aksel Celasun
Mobilnummer/cell phone: (+47) 900 15 103
Sentralbord/Support 4000 1850
ak...@abacus-it.nomailto:ak...@abacus-it.no
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Re: [asterisk-users] Error trying to add context: Context 'internal' tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri|*|*'
On Fri, Jun 18, 2010 at 7:44 AM, Aksel Celasun ak...@abacus-it.no wrote: Minor edit on the include = nighttime|12:30-8:00|mon-fri|*|* Correct now. This isn't how you do time based checks in asterisk. Lookup the application GotoIfTime. -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error trying to add context: Context 'internal' tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri|*|*'
On Friday 18 June 2010 09:49:39 Warren Selby wrote: On Fri, Jun 18, 2010 at 7:44 AM, Aksel Celasun ak...@abacus-it.no wrote: Minor edit on the include = nighttime|12:30-8:00|mon-fri|*|* Correct now. This isn't how you do time based checks in asterisk. Lookup the application GotoIfTime. Actually, it is an old method that still works, but as Warren mentioned, you should endeavor to switch to using GotoIfTime, as there's a nasty race condition inherent in using timed includes. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error trying to add context: Context 'internal' tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri|*|*'
Hi again Thank you Warren, GotoIfTime was the deal! And easy to use! Gr8. Best regards. Aksel Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne av Warren Selby Sendt: 18. juni 2010 16:50 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] Error trying to add context: Context 'internal' tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri|*|*' On Fri, Jun 18, 2010 at 7:44 AM, Aksel Celasun ak...@abacus-it.nomailto:ak...@abacus-it.no wrote: Minor edit on the include = nighttime|12:30-8:00|mon-fri|*|* Correct now. This isn't how you do time based checks in asterisk. Lookup the application GotoIfTime. -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error trying to add context: Context 'internal' tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri|*|*'
Thank you for the info. As I wrote to Warren GotoIfTime was easy to use and seemed more flexible, Got it working now! Perfect! Only one thing left now, and my system is pretty much ready for live testing, Surely easy for the user list, so it will come in another mail soon, after I have done Some more research. (how the receptionist can transfer calls to SIP extensions internally) Best regards Aksel -Opprinnelig melding- Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne av Tilghman Lesher Sendt: 18. juni 2010 18:01 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] Error trying to add context: Context 'internal' tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri|*|*' On Friday 18 June 2010 09:49:39 Warren Selby wrote: On Fri, Jun 18, 2010 at 7:44 AM, Aksel Celasun ak...@abacus-it.no wrote: Minor edit on the include = nighttime|12:30-8:00|mon-fri|*|* Correct now. This isn't how you do time based checks in asterisk. Lookup the application GotoIfTime. Actually, it is an old method that still works, but as Warren mentioned, you should endeavor to switch to using GotoIfTime, as there's a nasty race condition inherent in using timed includes. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] error message in CLI regarding SET Timeout
Hello! Does anybody know why i get the following error in CLI regarding the timeout option, when I dial sip ext 501? I get the message playing in the background, but the cli output confuses me. Running asterisk 1.6 on centos. == Using SIP RTP CoS mark 5 -- Executing [...@phones:1] Answer(SIP/301-0203, ) in new stack -- Executing [...@phones:2] Set(SIP/301-0203, TIMEOUT(5)=timeout) in new stack [Jun 17 10:35:31] ERROR[30876]: func_timeout.c:184 timeout_write: Unknown timeout type specified. -- Executing [...@phones:3] Set(SIP/301-0203, Timeout(response)=30) in new stack [Jun 17 10:35:31] ERROR[30876]: pbx.c:3386 ast_func_write: Function Timeout not registered -- Executing [...@phones:4] BackGround(SIP/301-0203, velkommen_abacustast123vent_) in new stack extensions.conf snipped. exten = 501,1,Answer exten = 501,n,Set(Timeout(5)=timeout) exten = 501,n,Set(Timeout(30)=response) exten = 501,n,Background(velkommen_abacustast123vent_) Med vennlig hilsen Abacus IT AS - din Visma Software Partner Tor Aksel Celasun Mobilnummer 900 15 103 Sentralbord/Support 4000 1850 ak...@abacus-it.nomailto:ak...@abacus-it.no -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error of FreePBX after installing from Yum Repository of Asterisk
I actually commented all this in safe_asterisk and now asterisk loads all
fine at the beginning. Is this okay to do? Also at the beginning of the file
I commented #TTY=9 as well. Can someone shade some light as to what TTY is
and how it can have an adverse effect if it's not available?
#if test x$TTY != x ; then
# if test -c /dev/tty${TTY} ; then
# TTY=tty${TTY}
# elif test -c /dev/vc/${TTY} ; then
# TTY=vc/${TTY}
# else
# message Cannot find specified TTY (${TTY})
# exit 1
# fi
# ASTARGS=${ASTARGS} -vvvg
# if test x$CONSOLE != xno ; then
# ASTARGS=${ASTARGS} -c
# fi
#fi
On Mon, Jun 7, 2010 at 8:12 PM, bruce bruce bruceb...@gmail.com wrote:
I did see the TTY=9 on the third or fourth line but commenting that doesn't
help much. I would really appreciate it if you can send the changes you
made.
Indeed it is a VPS.
Thanks,
Bruce
On Mon, Jun 7, 2010 at 7:49 PM, Warren Selby wcse...@selbytech.comwrote:
*chown: cannot access `/dev/tty9': No such file or directory*
I had this error on a VPS (virtual server) that did not have access to
tty's. You can take the TTY statement out of safe_asterisk script and then
try it again. I don't have the exact code right now because I'm on my
phone, but you should be able to find it if you read through that file.
Thanks,
--Warren Selby
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Re: [asterisk-users] Error of FreePBX after installing from Yum Repository of Asterisk
CentOS 5.4 and asterisk does stay running after it's loaded by asterisk -g. But the chkconfig --add asterisk doesn't work :( I need FOP and this error should go away as it's annoying. I don't see this on Trixbox, piaf, or Elastix. It shouldn't be on my install either. Thanks for the input. On Sun, Jun 6, 2010 at 11:43 PM, Seann Clark nombran...@tsukinokage.netwrote: The op_server.pl is part of the Flash Operators Panel, which isn't really important to the operation of the PBX, it is just a nice pretty interface showing what lines and what groups are active. What O/S are you using? Are there any errors in the asterisk logs? Does asterisk stay running after it starts? ~Seann On 6/6/2010 5:00 PM, bruce bruce wrote: Reboot like 10 times and the problem still presists. Also, upon reboot despite having done chkconfig --add asterisk asterisk still doesn't load automatically. And amportal start fails. So, I have to do asterisk -g first and then amportal start. Wondering if that might be related? Thanks for the input. On Sun, Jun 6, 2010 at 4:47 PM, dotnetdub dotnet...@gmail.com mailto: dotnet...@gmail.com wrote: On 6 June 2010 19:48, bruce bruce bruceb...@gmail.com mailto:bruceb...@gmail.com wrote: Hi Guys, Just did an Asterisk 1.6.x (repo install) and FreePBX (source install). When trying to dial a number, I get this: tel*CLI Use of uninitialized value in hash element at /var/www/html/panel/op_server.pl http://op_server.pl line 3367. Use of uninitialized value in concatenation (.) or string at /var/www/html/panel/op_server.pl http://op_server.pl line 3372. Use of uninitialized value in pattern match (m//) at /var/www/html/panel/op_server.pl http://op_server.pl line 3374. What could be causing that? I searched google and no useful information. Thanks, Bruce Reboot and should go away -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error of FreePBX after installing from Yum Repository of Asterisk
On Mon, 7 Jun 2010, bruce bruce wrote:
CentOS 5.4 and asterisk does stay running after it's loaded by asterisk
-g. But the chkconfig --add asterisk doesn't work :(
What does chkconfig --list asterisk show?
The add command looks in the asterisk script for a line that looks like:
# chkconfig: 2345 98 98
This says that chkconfig should create the appropriate links in the
/etc/rc{x}.d/ hierarchy so that Asterisk will be started at runlevels 2,
3, 4, 5 with a start priority of 98 (see man chkconfig for details) and
a stop priority of 98. Since CentOS servers should be running at runlevel
3, the 2, 4, and 5 are superfluous.
If there is no such line, chkconfig will not create the appropriate links.
Also, if /etc/init.d/asterisk does not have execute privileges, it will
not be executed on startup and Asterisk will not be running as expected.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Error of FreePBX after installing from Yum Repository of Asterisk
Steve Edwards wrote:
On Mon, 7 Jun 2010, bruce bruce wrote:
CentOS 5.4 and asterisk does stay running after it's loaded by asterisk
-g. But the chkconfig --add asterisk doesn't work :(
What does chkconfig --list asterisk show?
The add command looks in the asterisk script for a line that looks like:
# chkconfig: 2345 98 98
This says that chkconfig should create the appropriate links in the
/etc/rc{x}.d/ hierarchy so that Asterisk will be started at runlevels 2,
3, 4, 5 with a start priority of 98 (see man chkconfig for details) and
a stop priority of 98. Since CentOS servers should be running at runlevel
3, the 2, 4, and 5 are superfluous.
If there is no such line, chkconfig will not create the appropriate links.
Also, if /etc/init.d/asterisk does not have execute privileges, it will
not be executed on startup and Asterisk will not be running as expected.
I am running centos 5.4 myself. What I have for the chkconfig, mentioned
above is:
Mon Jun 07-12:29:27-r...@eiji.tsukinokage.net:cgi-bin chkconfig --list
asterisk
asterisk0:off 1:off 2:on3:on4:on5:on6:off
You should see the same thing if it is set up correctly. If it is in
there, try service asterisk start, and verify it is still running. If it
isn't, check /var/log/asterisk/messages and that should give you an idea
as to what killed it.
FOP is in an error error state is most likely due to channel's not being
found, and thus the pattern matches in the code, at the lines specified
are not matching anything and causing errors with the perl code that
runs the FOP server. This points back to the PBX's configuration. You
can look into what FOP is looking for in the op_server.cfg file and see
if your manager is allowing connections, etc, for the program to work.
My suggestion is make sure that asterisk by itself works in terms of
starting up cleaning, then verify amportal/FreePBX is configured and
working correctly, then FOP should work correctly after that.
Regards,
Seann
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Re: [asterisk-users] Error of FreePBX after installing from Yum Repository of Asterisk
Thanks for the input Seann and Steve. That is insightful. I did run
chkconfig --list asterisk and following is the output:
*[r...@tel ~]# chkconfig --list asterisk*
*asterisk0:off 1:off 2:on3:on4:on5:on6:off*
In file /usr/sbin/safe_asterisk I have priority for asterisk set at 9.
*# run asterisk with this priority*
*PRIORITY=9*
/var/log/messages doesn't show anything important or related to why asterisk
not starting at startup. I think asterisk should start first and then
amportal will start as well is asterisk does start.
Here is what happens if I do amportal restart:
*[r...@tel ~]# amportal restart*
*
*
*STOPPING ASTERISK*
*Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl
exist?)*
*Asterisk Stopped*
*
*
*STOPPING FOP SERVER*
*SETTING FILE PERMISSIONS*
*chown: cannot access `/dev/tty9': No such file or directory*
*Permissions OK*
*
*
*STARTING ASTERISK*
*Cannot find specified TTY (9)*
*safe_asterisk: no process killed*
*mpg123: no process killed*
*
*
*-*
*Asterisk could not start!*
*Use 'tail /var/log/asterisk/full' to find out why.*
*-*
*[r...@tel ~]#*
*[r...@tel ~]#*
*[r...@tel ~]# asterisk -g*
*[r...@tel ~]# amportal start*
*
*
*
*
*SETTING FILE PERMISSIONS*
*chown: cannot access `/dev/tty9': No such file or directory*
*Permissions OK*
*
*
*STARTING ASTERISK*
*Asterisk is already running*
*
*
*STARTING FOP SERVER*
*FOP Server Started*
I did a tail and here it is:
*[r...@tel ~]# tail /var/log/asterisk/full*
*[Jun 7 22:17:36] WARNING[4384] chan_dahdi.c: Ignoring any changes to
'userbase' (on reload) at line 23.*
*[Jun 7 22:17:36] WARNING[4384] chan_dahdi.c: Ignoring any changes to
'vmsecret' (on reload) at line 31.*
*[Jun 7 22:17:36] WARNING[4384] chan_dahdi.c: Ignoring any changes to
'hassip' (on reload) at line 35.*
*[Jun 7 22:17:36] WARNING[4384] chan_dahdi.c: Ignoring any changes to
'hasiax' (on reload) at line 39.*
*[Jun 7 22:17:36] WARNING[4384] chan_dahdi.c: Ignoring any changes to
'hasmanager' (on reload) at line 47.*
*[Jun 7 22:17:36] NOTICE[4384] chan_skinny.c: Configuring skinny from
skinny.conf*
*[Jun 7 22:17:36] WARNING[4384] res_musiconhold.c: Cannot open dir
/var/lib/asterisk/moh or dir does not exist*
*[Jun 7 22:17:36] WARNING[4384] res_musiconhold.c: Cannot open dir
/var/lib/asterisk/moh/.nomusic_reserved or dir does not exist*
*[Jun 7 22:17:36] WARNING[4384] res_musiconhold.c: No music on hold classes
configured, disabling music on hold.*
*[Jun 7 22:18:40] ERROR[4479] pbx.c: Did not remove this priority label
(57/vmxopts) from the peer_label_table of context macro-vm, extension vmx!*
Thanks,
Bruce
On Mon, Jun 7, 2010 at 3:29 PM, Steve Edwards asterisk@sedwards.comwrote:
On Mon, 7 Jun 2010, bruce bruce wrote:
CentOS 5.4 and asterisk does stay running after it's loaded by asterisk
-g. But the chkconfig --add asterisk doesn't work :(
What does chkconfig --list asterisk show?
The add command looks in the asterisk script for a line that looks like:
# chkconfig: 2345 98 98
This says that chkconfig should create the appropriate links in the
/etc/rc{x}.d/ hierarchy so that Asterisk will be started at runlevels 2,
3, 4, 5 with a start priority of 98 (see man chkconfig for details) and
a stop priority of 98. Since CentOS servers should be running at runlevel
3, the 2, 4, and 5 are superfluous.
If there is no such line, chkconfig will not create the appropriate links.
Also, if /etc/init.d/asterisk does not have execute privileges, it will
not be executed on startup and Asterisk will not be running as expected.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Error of FreePBX after installing from Yum Repository of Asterisk
On 6/7/2010 5:20 PM, bruce bruce wrote:
Thanks for the input Seann and Steve. That is insightful. I did run
chkconfig --list asterisk and following is the output:
*[r...@tel ~]# chkconfig --list asterisk*
*asterisk0:off 1:off 2:on3:on4:on5:on6:off*
In file /usr/sbin/safe_asterisk I have priority for asterisk set at 9.
*# run asterisk with this priority*
*PRIORITY=9*
/var/log/messages doesn't show anything important or related to why
asterisk not starting at startup. I think asterisk should start first
and then amportal will start as well is asterisk does start.
Here is what happens if I do amportal restart:
*[r...@tel ~]# amportal restart*
*
*
*STOPPING ASTERISK*
*Unable to connect to remote asterisk (does
/var/run/asterisk/asterisk.ctl exist?)*
*Asterisk Stopped*
*
*
*STOPPING FOP SERVER*
*SETTING FILE PERMISSIONS*
*chown: cannot access `/dev/tty9': No such file or directory*
*Permissions OK*
*
*
*STARTING ASTERISK*
*Cannot find specified TTY (9)*
*safe_asterisk: no process killed*
*mpg123: no process killed*
*
*
*-*
*Asterisk could not start!*
*Use 'tail /var/log/asterisk/full' to find out why.*
*-*
*[r...@tel ~]#*
*[r...@tel ~]#*
*[r...@tel ~]# asterisk -g*
*[r...@tel ~]# amportal start*
*
*
*
*
*SETTING FILE PERMISSIONS*
*chown: cannot access `/dev/tty9': No such file or directory*
*Permissions OK*
*
*
*STARTING ASTERISK*
*Asterisk is already running*
*
*
*STARTING FOP SERVER*
*FOP Server Started*
I did a tail and here it is:
*[r...@tel ~]# tail /var/log/asterisk/full*
*[Jun 7 22:17:36] WARNING[4384] chan_dahdi.c: Ignoring any changes to
'userbase' (on reload) at line 23.*
*[Jun 7 22:17:36] WARNING[4384] chan_dahdi.c: Ignoring any changes to
'vmsecret' (on reload) at line 31.*
*[Jun 7 22:17:36] WARNING[4384] chan_dahdi.c: Ignoring any changes to
'hassip' (on reload) at line 35.*
*[Jun 7 22:17:36] WARNING[4384] chan_dahdi.c: Ignoring any changes to
'hasiax' (on reload) at line 39.*
*[Jun 7 22:17:36] WARNING[4384] chan_dahdi.c: Ignoring any changes to
'hasmanager' (on reload) at line 47.*
*[Jun 7 22:17:36] NOTICE[4384] chan_skinny.c: Configuring skinny from
skinny.conf*
*[Jun 7 22:17:36] WARNING[4384] res_musiconhold.c: Cannot open dir
/var/lib/asterisk/moh or dir does not exist*
*[Jun 7 22:17:36] WARNING[4384] res_musiconhold.c: Cannot open dir
/var/lib/asterisk/moh/.nomusic_reserved or dir does not exist*
*[Jun 7 22:17:36] WARNING[4384] res_musiconhold.c: No music on hold
classes configured, disabling music on hold.*
*[Jun 7 22:18:40] ERROR[4479] pbx.c: Did not remove this priority
label (57/vmxopts) from the peer_label_table of context macro-vm,
extension vmx!*
Thanks,
Bruce
On Mon, Jun 7, 2010 at 3:29 PM, Steve Edwards asterisk.org
http://asterisk.org@sedwards.com http://sedwards.com wrote:
On Mon, 7 Jun 2010, bruce bruce wrote:
CentOS 5.4 and asterisk does stay running after it's loaded by
asterisk
-g. But the chkconfig --add asterisk doesn't work :(
What does chkconfig --list asterisk show?
The add command looks in the asterisk script for a line that
looks like:
# chkconfig: 2345 98 98
This says that chkconfig should create the appropriate links in the
/etc/rc{x}.d/ hierarchy so that Asterisk will be started at
runlevels 2,
3, 4, 5 with a start priority of 98 (see man chkconfig for
details) and
a stop priority of 98. Since CentOS servers should be running at
runlevel
3, the 2, 4, and 5 are superfluous.
If there is no such line, chkconfig will not create the
appropriate links.
Also, if /etc/init.d/asterisk does not have execute privileges, it
will
not be executed on startup and Asterisk will not be running as
expected.
--
Thanks in advance,
-
First, I would create the directories that it is missing, and view your
tty's in /dev (ls -Al /dev | grep tty) and validate it is there, and
what permissions it has. Mine, default install, has:
crw-rw 1 root tty 4, 9 Jun 7 17:24 tty9
~Seann
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Re: [asterisk-users] Error of FreePBX after installing from Yum Repository of Asterisk
chown: cannot access `/dev/tty9': No such file or directory I had this error on a VPS (virtual server) that did not have access to tty's. You can take the TTY statement out of safe_asterisk script and then try it again. I don't have the exact code right now because I'm on my phone, but you should be able to find it if you read through that file. Thanks, --Warren Selby-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error of FreePBX after installing from Yum Repository of Asterisk
I did see the TTY=9 on the third or fourth line but commenting that doesn't help much. I would really appreciate it if you can send the changes you made. Indeed it is a VPS. Thanks, Bruce On Mon, Jun 7, 2010 at 7:49 PM, Warren Selby wcse...@selbytech.com wrote: *chown: cannot access `/dev/tty9': No such file or directory* I had this error on a VPS (virtual server) that did not have access to tty's. You can take the TTY statement out of safe_asterisk script and then try it again. I don't have the exact code right now because I'm on my phone, but you should be able to find it if you read through that file. Thanks, --Warren Selby -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error of FreePBX after installing from Yum Repository of Asterisk
Hi Guys, Just did an Asterisk 1.6.x (repo install) and FreePBX (source install). When trying to dial a number, I get this: tel*CLI Use of uninitialized value in hash element at /var/www/html/panel/ op_server.pl line 3367. Use of uninitialized value in concatenation (.) or string at /var/www/html/panel/op_server.pl line 3372. Use of uninitialized value in pattern match (m//) at /var/www/html/panel/ op_server.pl line 3374. What could be causing that? I searched google and no useful information. Thanks, Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error of FreePBX after installing from Yum Repository of Asterisk
On 6 June 2010 19:48, bruce bruce bruceb...@gmail.com wrote: Hi Guys, Just did an Asterisk 1.6.x (repo install) and FreePBX (source install). When trying to dial a number, I get this: tel*CLI Use of uninitialized value in hash element at /var/www/html/panel/ op_server.pl line 3367. Use of uninitialized value in concatenation (.) or string at /var/www/html/panel/op_server.pl line 3372. Use of uninitialized value in pattern match (m//) at /var/www/html/panel/ op_server.pl line 3374. What could be causing that? I searched google and no useful information. Thanks, Bruce Reboot and should go away -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error of FreePBX after installing from Yum Repository of Asterisk
Reboot like 10 times and the problem still presists. Also, upon reboot despite having done chkconfig --add asterisk asterisk still doesn't load automatically. And amportal start fails. So, I have to do asterisk -g first and then amportal start. Wondering if that might be related? Thanks for the input. On Sun, Jun 6, 2010 at 4:47 PM, dotnetdub dotnet...@gmail.com wrote: On 6 June 2010 19:48, bruce bruce bruceb...@gmail.com wrote: Hi Guys, Just did an Asterisk 1.6.x (repo install) and FreePBX (source install). When trying to dial a number, I get this: tel*CLI Use of uninitialized value in hash element at /var/www/html/panel/op_server.pl line 3367. Use of uninitialized value in concatenation (.) or string at /var/www/html/panel/op_server.pl line 3372. Use of uninitialized value in pattern match (m//) at /var/www/html/panel/ op_server.pl line 3374. What could be causing that? I searched google and no useful information. Thanks, Bruce Reboot and should go away -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error of FreePBX after installing from Yum Repository of Asterisk
The op_server.pl is part of the Flash Operators Panel, which isn't really important to the operation of the PBX, it is just a nice pretty interface showing what lines and what groups are active. What O/S are you using? Are there any errors in the asterisk logs? Does asterisk stay running after it starts? ~Seann On 6/6/2010 5:00 PM, bruce bruce wrote: Reboot like 10 times and the problem still presists. Also, upon reboot despite having done chkconfig --add asterisk asterisk still doesn't load automatically. And amportal start fails. So, I have to do asterisk -g first and then amportal start. Wondering if that might be related? Thanks for the input. On Sun, Jun 6, 2010 at 4:47 PM, dotnetdub dotnet...@gmail.com mailto:dotnet...@gmail.com wrote: On 6 June 2010 19:48, bruce bruce bruceb...@gmail.com mailto:bruceb...@gmail.com wrote: Hi Guys, Just did an Asterisk 1.6.x (repo install) and FreePBX (source install). When trying to dial a number, I get this: tel*CLI Use of uninitialized value in hash element at /var/www/html/panel/op_server.pl http://op_server.pl line 3367. Use of uninitialized value in concatenation (.) or string at /var/www/html/panel/op_server.pl http://op_server.pl line 3372. Use of uninitialized value in pattern match (m//) at /var/www/html/panel/op_server.pl http://op_server.pl line 3374. What could be causing that? I searched google and no useful information. Thanks, Bruce Reboot and should go away -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME Cryptographic Signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error compiling DAHDI...
The resolution [1] to this issue was to uninstall and reinstall [2] the kernel headers on the machine...just in case anyone else runs into this issue and would like to know how it was solved. [1] https://issues.asterisk.org/view.php?id=17411 [2] run these commands to reinstall kernel headers: ]# yum remove kernel-devel ]# yum install kernel-devel Thanks, --Warren Selby On May 25, 2010, at 9:48 PM, Warren Selby wcse...@selbytech.com wrote: I was at a client site tonight to install OSLEC on his machine running asterisk 1.6.0.22 and DAHDI 2.2.1 installed via yum. I stopped asterisk and DAHDI, downloaded the latest version of DAHDI 2.2.1 (dahdi-linux-complete-2.2.1.2+2.2.1.1) and made the necessary changes to compile OSLEC with DAHDI, but I ran into compilation issues that I had never seen before. So as a test I deleted my /usr/ src/dahdi/ directory, re-expanded my tarball (so that I had a vanilla DAHDI package), and tried to compile it again, and I got the same errors. I have not seen these errors before, and I'm not sure what would cause them. Can anyone help shed some light on this? The 'make' output: *snip* -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error compiling DAHDI...
On Thu, Jun 3, 2010 at 5:02 PM, Warren Selby wcse...@selbytech.com wrote: The resolution [1] to this issue was to uninstall and reinstall [2] the kernel headers on the machine...just in case anyone else runs into this issue and would like to know how it was solved. [1] https://issues.asterisk.org/view.php?id=17411 [2] run these commands to reinstall kernel headers: ]# yum remove kernel-devel ]# yum install kernel-devel FYI: I just had the same problem with Asterisk 1.6 on FreeBSD 7.0 RELEASE. Even with correct headers it would complain about som opt_netgraph.h that never existed. Finally, I resorted to try with Asterisk 1.4 and it compiled fine. Conclusion: 1.6 won't work with FBSD 7.0 Best, Alejandro Imass Thanks, --Warren Selby On May 25, 2010, at 9:48 PM, Warren Selby wcse...@selbytech.com wrote: I was at a client site tonight to install OSLEC on his machine running asterisk 1.6.0.22 and DAHDI 2.2.1 installed via yum. I stopped asterisk and DAHDI, downloaded the latest version of DAHDI 2.2.1 (dahdi-linux-complete-2.2.1.2+2.2.1.1) and made the necessary changes to compile OSLEC with DAHDI, but I ran into compilation issues that I had never seen before. So as a test I deleted my /usr/src/dahdi/ directory, re-expanded my tarball (so that I had a vanilla DAHDI package), and tried to compile it again, and I got the same errors. I have not seen these errors before, and I'm not sure what would cause them. Can anyone help shed some light on this? The 'make' output: *snip* -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error compiling DAHDI...
I was at a client site tonight to install OSLEC on his machine running asterisk 1.6.0.22 and DAHDI 2.2.1 installed via yum. I stopped asterisk and DAHDI, downloaded the latest version of DAHDI 2.2.1 (dahdi-linux-complete-2.2.1.2+2.2.1.1) and made the necessary changes to compile OSLEC with DAHDI, but I ran into compilation issues that I had never seen before. So as a test I deleted my /usr/src/dahdi/ directory, re-expanded my tarball (so that I had a vanilla DAHDI package), and tried to compile it again, and I got the same errors. I have not seen these errors before, and I'm not sure what would cause them. Can anyone help shed some light on this? The 'make' output: dahdi]# make make -C drivers/dahdi/firmware firmware-loaders make[1]: Entering directory `/usr/src/dahdi/drivers/dahdi/ firmware' make[1]: Leaving directory `/usr/src/dahdi/drivers/dahdi/firmware' make -C /lib/modules/2.6.18-164.11.1.el5/build SUBDIRS=/usr/src/dahdi/drivers/dahdi DAHDI_INCLUDE=/usr/src/dahdi/include DAHDI_MODULES_EXTRA= HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m make[1]: Entering directory `/usr/src/kernels/2.6.18-164.11.1.el5-i686' CC [M] /usr/src/dahdi/drivers/dahdi/dahdi-base.o In file included from include/linux/spinlock.h:8, from include/linux/capability.h:45, from include/linux/sched.h:44, from include/linux/module.h:9, from /usr/src/dahdi/drivers/dahdi/dahdi-base.c:40: include/linux/config.h:6:2: warning: #warning Including config.h is deprecated. In file included from include/linux/spinlock.h:39, from include/linux/capability.h:45, from include/linux/sched.h:44, from include/linux/module.h:9, from /usr/src/dahdi/drivers/dahdi/dahdi-base.c:40: include/asm/spinlock.h:60: error: expected â)â before â*â token include/asm/spinlock.h:71: error: expected â)â before â*â token include/asm/spinlock.h:77: error: expected â)â before â*â token include/asm/spinlock.h:115: error: expected â)â before â*â token include/asm/spinlock.h:162: error: expected â)â before â*â token include/asm/spinlock.h:167: error: expected â)â before â*â token include/asm/spinlock.h:172: error: expected â)â before â*â token include/asm/spinlock.h:182: error: expected â)â before â*â token include/asm/spinlock.h:191: error: expected â)â before â*â token include/asm/spinlock.h:196: error: expected â)â before â*â token In file included from include/linux/capability.h:45, from include/linux/sched.h:44, from include/linux/module.h:9, from /usr/src/dahdi/drivers/dahdi/dahdi-base.c:40: include/linux/spinlock.h:43: error: expected â)â before â*â token include/linux/spinlock.h:44: error: expected â)â before â*â token include/linux/spinlock.h:45: error: expected â)â before â*â token include/linux/spinlock.h:46: error: expected â)â before â*â token include/linux/spinlock.h:47: error: expected â)â before â*â token include/linux/spinlock.h:48: error: expected â)â before â*â token include/linux/spinlock.h:49: error: expected â)â before â*â token include/linux/spinlock.h:50: error: expected â)â before â*â token include/linux/spinlock.h:51: error: expected â)â before â*â token include/linux/spinlock.h:52: error: expected â)â before â*â token include/linux/spinlock.h:53: error: expected â)â before â*â token include/linux/spinlock.h:54: error: expected â)â before â*â token include/linux/spinlock.h:55: error: expected â)â before â*â token include/linux/spinlock.h:56: error: expected â)â before â*â token include/linux/spinlock.h:57: error: expected â)â before â*â token include/linux/spinlock.h:58: error: expected â)â before â*â token include/linux/spinlock.h:59: error: expected â)â before â*â token include/linux/spinlock.h:60: error: expected â)â before â*â token include/linux/spinlock.h:61: error: expected â)â before â*â token include/linux/spinlock.h:62: error: expected â)â before â*â token include/linux/spinlock.h:63: error: expected â)â before â*â token include/linux/spinlock.h:64: error: expected â)â before â*â token include/linux/spinlock.h:65: error: expected â)â before â*â token include/linux/spinlock.h:66: error: expected â)â before â*â token include/linux/spinlock.h:67: error: expected â)â before â*â token include/linux/spinlock.h:68: error: expected â)â before â*â token include/linux/spinlock.h:69: error: expected â)â before â*â token include/linux/spinlock.h:70: error: expected â)â before â*â token include/linux/spinlock.h:477: error: expected declaration specifiers or â...â before âspinlock_tâ In file included from include/linux/time.h:7, from include/linux/timex.h:57, from include/linux/sched.h:48, from include/linux/module.h:9, from /usr/src/dahdi/drivers/dahdi/dahdi-base.c:40: include/linux/seqlock.h:34: error: expected specifier-qualifier-list before âspinlock_tâ include/linux/seqlock.h: In
Re: [asterisk-users] Error at start of asterisk with cdr_addon_mysql.o
On Wed, 2010-05-12 at 22:10 -0700, Steve Edwards wrote:
On Thu, 13 May 2010, Pham Quy wrote:
Hi all,
I use asterisk-1.6.2.7 and asterisk addon version 1.6.2.1.
It started ok with out cdr_addon_mysql.o. But when I put
cdr_addon_mysql.o in to modules folder, it fail at start and the
following out has been thrown:
--
[r...@localhost modules]# /usr/sbin/safe_asterisk: line 145: 13270
Segmentation fault (core dumped) nice -n $PRIORITY ${ASTBINDIR}/asterisk
-f ${CLIARGS} ${ASTARGS} /dev/${TTY} 21 /dev/${TTY}
Asterisk exited with exit status 139
Asterisk exited on signal 11
Automatically restarting Asterisk.
---
What is the problem?
The problem is...
You have no clue[s] :)
First off, the module should be cdr_addon_mysql.so, not cdr_addon_mysql.o.
If you don't have the so in /usr/lib/asterisk/modules/ something is
wrong with your build.
Try something like this:
sudo -u whatever-user-runs-asterisk-on-your-system\
/usr/sbin/asterisk -c -d -d -d -f -g -n -v -v -v
Or, you can start Asterisk without loading cdr_addon_mysql.so and then
load it from the Asterisk CLI. It sounds like you are auto-loading modules
so you could add noload=cdr_addon_mysql.so to
/etc/asterisk/modules.conf to get Asterisk running and then load it with
something like load cdr_addon_mysql.so
I'm a 1.2 Luddite so the commands may have changed slightly. Also,
depending on the specifics of your installation, the paths may be
different.
See if this gives you any clues.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
Hi,
Yes, it was a typing mistake, i meant cdr_addon_mysql.so. After manually
loadind the module, it turn out there is an mistake in my cdr_mysql.conf
I fixed it and everything work fine.
Thanks.
Quyps
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[asterisk-users] Error at start of asterisk with cdr_addon_mysql.o
Hi all,
I use asterisk-1.6.2.7 and asterisk addon version 1.6.2.1.
It started ok with out cdr_addon_mysql.o. But when I put
cdr_addon_mysql.o in to modules folder, it fail at start and the
following out has been thrown:
--
[r...@localhost modules]# /usr/sbin/safe_asterisk: line 145: 13270
Segmentation fault (core dumped) nice -n $PRIORITY ${ASTBINDIR}/asterisk
-f ${CLIARGS} ${ASTARGS} /dev/${TTY} 21 /dev/${TTY}
Asterisk exited with exit status 139
Asterisk exited on signal 11
Automatically restarting Asterisk.
---
What is the problem?
Thanks in advance.
Quyps
--
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Re: [asterisk-users] Error at start of asterisk with cdr_addon_mysql.o
On Thu, 13 May 2010, Pham Quy wrote:
Hi all,
I use asterisk-1.6.2.7 and asterisk addon version 1.6.2.1.
It started ok with out cdr_addon_mysql.o. But when I put
cdr_addon_mysql.o in to modules folder, it fail at start and the
following out has been thrown:
--
[r...@localhost modules]# /usr/sbin/safe_asterisk: line 145: 13270
Segmentation fault (core dumped) nice -n $PRIORITY ${ASTBINDIR}/asterisk
-f ${CLIARGS} ${ASTARGS} /dev/${TTY} 21 /dev/${TTY}
Asterisk exited with exit status 139
Asterisk exited on signal 11
Automatically restarting Asterisk.
---
What is the problem?
The problem is...
You have no clue[s] :)
First off, the module should be cdr_addon_mysql.so, not cdr_addon_mysql.o.
If you don't have the so in /usr/lib/asterisk/modules/ something is
wrong with your build.
Try something like this:
sudo -u whatever-user-runs-asterisk-on-your-system\
/usr/sbin/asterisk -c -d -d -d -f -g -n -v -v -v
Or, you can start Asterisk without loading cdr_addon_mysql.so and then
load it from the Asterisk CLI. It sounds like you are auto-loading modules
so you could add noload=cdr_addon_mysql.so to
/etc/asterisk/modules.conf to get Asterisk running and then load it with
something like load cdr_addon_mysql.so
I'm a 1.2 Luddite so the commands may have changed slightly. Also,
depending on the specifics of your installation, the paths may be
different.
See if this gives you any clues.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
--
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Re: [asterisk-users] Error redirecting an incoming call of a SIP provider to a local extension
On 20/02/2010 01:35, Daniel Bareiro wrote: alderamin*CLI -- Executing [...@from-internal:1] Dial(SIP/danib-089f8820, SIP/300|30|tTrm) in new stack [Feb 19 19:22:50] WARNING[19254]: app_dial.c:1237 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) Well, looks like your * server is simply unable to dial the sip user '300'. Either there is some call-limit in place, or problem with the registration of the phone ? It is probable that this can be due to a problem of interaction between contexts? I copy the content of extensions.conf and sip.conf to see if it can help to find the problem: What could be of some use, is the result of sip show peer 300 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error redirecting an incoming call of a SIP provider to a local extension
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all!
I am trying to redirect to a local extension the incoming calls that I
receive to an account which I have in iptel.org, but when receiving I'm
obtaining this error:
alderamin*CLI
-- Executing [...@from-internal:1] Dial(SIP/danib-089f8820,
SIP/300|30|tTrm) in new stack
[Feb 19 19:22:50] WARNING[19254]: app_dial.c:1237 dial_exec_full: Unable
to create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
[Feb 19 19:23:00] WARNING[19254]: pbx.c:2529 __ast_pbx_run: Timeout, but
no rule 't' in context 'from-internal'
It is probable that this can be due to a problem of interaction between
contexts? I copy the content of extensions.conf and sip.conf to see if
it can help to find the problem:
-
extensions.conf:
; DGB - 20091114
[general]
autofallthrough=no
[macro-dial]
exten = s,1,Dial(${ARG1},15)
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(${macro_ext...@voicemail,u)
exten = s-NOANSWER,n,Hangup
exten = s-BUSY,1,Voicemail(${macro_ext...@voicemail,b)
exten = s-BUSY,n,Hangup
exten = s-CHANUNAVAIL,1,Playback(pbx-invalid)
[from-internal]
; Llamadas a extensiones SIP
exten = _2xx,1,Macro(dial,SIP/${EXTEN})
exten = _2xx,n,Hangup
exten = 300,1,Dial(SIP/300,30,tTrm)
; Extension analogica
exten = 402,1,Macro(dial,DAHDI/2)
exten = 402,n,Hangup
; Directorio de extensiones
exten = *400,1,Directory(voicemail,from-internal)
; Musica en espera
exten = *300,1,Answer
exten = *300,n,SetMusicOnHold(default)
exten = *300,n,WaitMusicOnHold(2000)
exten = *300,n,Hangup
; Prueba de Eco
exten = *200,1,Answer
exten = *200,n,Playback(demo-echotest)
exten = *200,n,Echo
exten = *200,n,Playback(demo-echodone)
exten = *200,n,Hangup
; Acceso a voicemail
exten = *100,1,Answer
exten = *100,n,Wait(1)
exten = *100,n,VoiceMailMain(${CALLERID(num)}...@voicemail)
exten = *100,n,Hangup
; Llamadas salientes
exten = _9.,1,Dial(DAHDI/1/${EXTEN:1})
exten = _9.,n,Hangup
; Call a number at iptel.org
exten = _0.,1,Dial(SIP/iptel/${EXTEN:1},20,r))
exten = _0.,n,Hangup
[from-pstn]
; incoming calls from FXO port are directed to this context
exten = s,1,Answer()
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=15)
exten = s,n,Background(contestador1)
exten = i,1,Goto(from-pstn,s,1)
exten = t,1,Playback(locomunicoconelinterno1)
exten = t,n,Dial(SIP/200,25)
exten = t,n,VoiceMail(2...@voicemail,20)
exten = t,n,Hangup()
include = from-internal
-
sip.conf:
[general]
[...]
; register with iptel.org
register = danib:mlrzv...@iptel.org/300
[...]
; Outgoing to iptel.org
[iptel]
type=friend
username=danib
secret=myspasswd
host=iptel.org
canreinvite=no
qualify=300
insecure=port,invite ; required for incoming ekiga.net calls
context = from-internal
-
Thanks in advance for your replies.
Regards,
Daniel
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[asterisk-users] Error and call drops
Hi, does anyone have an info into what could cause [Nov 28 14:24:48] ERROR[11964] utils.c: write() returned error: Broken pipe [Nov 28 14:25:08] ERROR[12540] utils.c: write() returned error: Broken pipe [Nov 28 14:25:08] ERROR[12540] utils.c: write() returned error: Broken pipe [Nov 28 14:26:23] ERROR[13098] utils.c: write() returned error: Broken pipe Is it a write process or a problem with one of the scripts I am running? I am seeing this over and over again and experience call drops on a percentage of calls. Thanks Lee -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error and call drops
I've found that I get this error when I don't properly listen for asterisk responses to my commands in my agi scripts. Anytime you send a command to asterisk from an agi script, asterisk sends a response to the script with the result of the command (i.e a 200 ok response if asterisk was able to properly execute the command). If your script doesn't properly handle these responses, you get the error mentioned below. It's never caused any of my calls to drop, though. Try turning on AGI debug to see if this is the case for you. Thanks, --Warren Selby On Jan 26, 2010, at 5:11 AM, Lee Archer lee.arc...@thebigword.com wrote: Hi, does anyone have an info into what could cause [Nov 28 14:24:48] ERROR[11964] utils.c: write() returned error: Broken pipe [Nov 28 14:25:08] ERROR[12540] utils.c: write() returned error: Broken pipe [Nov 28 14:25:08] ERROR[12540] utils.c: write() returned error: Broken pipe [Nov 28 14:26:23] ERROR[13098] utils.c: write() returned error: Broken pipe Is it a write process or a problem with one of the scripts I am running? I am seeing this over and over again and experience call drops on a percentage of calls. Thanks Lee -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error and call drops
Hi, does anyone have an info into what could cause [Nov 28 14:24:48] ERROR[11964] utils.c: write() returned error: Broken pipe I have had the same issue with a PHP script that logs into the manager interface. If you don't wait for the AMI response, then log off before closing the connection you will get those errors. I also haven't seen any call drops. I would urge you to check your scripts, and put some 2 second waits before a logoff and closing the socket and see if that helps. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] error compile dahdi with latest kernels.
hello, all of users: there are header files missed when you compile dahdi with kernel-2.6.29 or 2.6.33. i believe that few files are affected: wctdm.c dahdi-base.c wcb4xxp/base.c, opvxa1200.c... the errors look like these: from /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c:61: /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/include/dahdi/dahdi_config.h:27:28: error: linux/autoconf.h: No such file or directory /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c: In function '__qevent': /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c:839: error: 'TASK_INTERRUPTIBLE' undeclared (first use in this function) /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c:839: error: (Each undeclared identifier is reported only once /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c:839: error: for each function it appears in.) /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c: In function 'schluffen': /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c:867: error: dereferencing pointer to incomplete type /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c:867: error: 'TASK_INTERRUPTIBLE' undeclared (first use in this function) /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c:869: error: implicit declaration of function 'signal_pending' /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c:870: error: implicit declaration of function 'schedule' /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c:872: error: dereferencing pointer to incomplete type /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c:872: error: 'TASK_RUNNING' undeclared (first use in this function) /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c: In function 'dahdi_timer_ioctl': /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c:3418: error: 'TASK_INTERRUPTIBLE' undeclared (first use in this function) /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c: In function 'dahdi_chanandpseudo_ioctl': /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c:4419: error: 'TASK_INTERRUPTIBLE' undeclared (first use in this function) /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c: In function '__dahdi_getbuf_chunk': /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c:6075: error: 'TASK_INTERRUPTIBLE' undeclared (first use in this function) /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c: In function '__rbs_otimer_expire': /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c:6263: error: 'TASK_INTERRUPTIBLE' undeclared (first use in this function) /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c: In function '__putbuf_chunk': /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c:7203: error: 'TASK_INTERRUPTIBLE' undeclared (first use in this function) /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c: In function 'dahdi_hdlc_finish': == after digging the code, i changed the files and add some linux headers. #include linux/kernel.h #include linux/errno.h +#include linux/sched.h #include linux/module.h #include linux/proc_fs.h = and add this: #ifdef __KERNEL__ #include linux/version.h #if LINUX_VERSION_CODE KERNEL_VERSION(2,6,18) #include linux/config.h #else +#include generated/autoconf.h -#include linux/autoconf.h #endif #endif = Regards! zhulizhong ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error compile dahdi with latest kernels.
On Thu, Jan 07, 2010 at 04:19:21PM +0800, james.zhu wrote: hello, all of users: there are header files missed when you compile dahdi with kernel-2.6.29 or 2.6.33. i believe that few files are affected: wctdm.c dahdi-base.c wcb4xxp/base.c, opvxa1200.c... the errors look like these: from /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/drivers/dahdi/dahdi-base.c:61: /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0/linux/include/dahdi/dahdi_config.h:27:28: error: linux/autoconf.h: No such file or directory http://svnview.digium.com/svn/dahdi?view=revisionrevision=7732 -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
