Re: [asterisk-users] SNMP monitoring
Hi am somehow interested in this topic. What did you want to use the snmp for,do u want to use for monitoring? I will be expecting reply from the house . Thanks On Feb 15, 2008 12:42 PM, Adrian Marsh [EMAIL PROTECTED] wrote: Thanks guys, On two cloned machines, on one I tried: yum install lm_sensors-devel bzip2-devel (ignoring newt, and these were the only ones missing) ..and it compiled ok. Then on the other I just added lm_sensors-devel and the configure –with-net-snmp worked ok, but it didn't compile the snmp module (but didn't complain either). So then I added bzip2-devel and all was well on the second machine (so both needed). So now the res_snmp.so module is loaded. I'll continue to work out what else is needed (I've no res_snmp.conf file, or net-snmp config updates done yet). Adrian *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Ricardo Carvalho *Sent:* 15 February 2008 00:29 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] SNMP monitoring Maybe you'r right and newt isn't really necessary. I just read somewhere that those dependencies were needed, I've installed them and it worked... Try to only install the other ones and if res_snmp gets compiled without it, great! Regards, Ricardo Carvalho. On Fri, Feb 15, 2008 at 12:01 AM, Darrick Hartman (lists) [EMAIL PROTECTED] wrote: Ricardo Carvalho wrote: I had the same problem some time ago... You got to install also this packages: net-snmp-devel newt-devel lm_sensors-devel bzip2-devel That should do it! Why would this depend on newt? net-snmp and lm-sensor headers and libraries make sense. newt doesn't make any sense as a dependency. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ -- 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 ___ -- 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] All trunk are busy please try your call again later
Good Day Find attached the relevant portions of the asterisk CLI. Please,which portion of the extension .conf should i send ? It is connected via RJ 45 connector to an E1 modem to the telco company. I use E1 link. I will appreciate your reply. Best Regards On Dec 18, 2007 4:02 PM, dave cantera [EMAIL PROTECTED] wrote: lolu, sounds more like a telco/itsp problem then *. I would tcpdump -i eth0 port 5060 to make sure it is actually going out... change 5060 if you have changed your port to your itsp, of course. to see what is going on as well as the other debugging notes mentioned in this thread. daveC Lolu Gbenga wrote: Good Day all Please I am having some issues on my voip asterisk server I make internal calls on extensions configured ie extension 192 can call extension 195 etc But each time i try to make calls outside the extension ie calling a GSM or an external line ,i always hear this response all trunk calls are busy please try your call again later Please how can i resolve this problem . I will appreciate your response. Best Regards Success ___ --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 -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --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 SIP SHOW PEERS Name/username HostDyn Nat ACL Port Status 7871/7871 (Unspecified)D 0Unmonitored ... ... 7874/7874 (Unspecified)D 0Unmonitored 108 sip peers [108 online , 0 offline] Verbosity is at least 3 ZAP SHOW CHANNELS Chan Extension Context Language MusicOnHold pseudodefault en 1default en 2default en ZAP SHOW CHANNELS Description Alarms IRQbpviol CRC4 T4XXP (PCI) Card 0 Span 1OK 0 0 0 T4XXP (PCI) Card 0 Span 2UNCONFIGUR 0 0 0 T4XXP (PCI) Card 0 Span 3UNCONFIGUR 0 0 0 T4XXP (PCI) Card 0 Span 4UNCONFIGUR 0 0 0 ZTDUMMY/1 1 UNCONFIGUR 0 0 0 Verbosity is at least 3 ___ --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] All trunk are busy please try your call again later
Thanks Please am using putty to again access to my Linux asterisk box. How can i use tcpdump to get your request on the exact Ethernet port and port number. I will appreciate your reply. Best Regards On Dec 18, 2007 4:02 PM, dave cantera [EMAIL PROTECTED] wrote: lolu, sounds more like a telco/itsp problem then *. I would tcpdump -i eth0 port 5060 to make sure it is actually going out... change 5060 if you have changed your port to your itsp, of course. to see what is going on as well as the other debugging notes mentioned in this thread. daveC Lolu Gbenga wrote: Good Day all Please I am having some issues on my voip asterisk server I make internal calls on extensions configured ie extension 192 can call extension 195 etc But each time i try to make calls outside the extension ie calling a GSM or an external line ,i always hear this response all trunk calls are busy please try your call again later Please how can i resolve this problem . I will appreciate your response. Best Regards Success ___ --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 -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --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] All trunk are busy please try your call again later
Hi Steve Am connected to the telco through an E1 link using modem(Watson 5 modem SDHSL 1 PAIR schmid telecommunications).The MODEM is connected to the asterisk box through RJ 45 to the asterisk box end and serial connector to the modem end . Which portion of the extension conf should i post ? Thanks On Dec 18, 2007 12:03 PM, Steve Totaro [EMAIL PROTECTED] wrote: Lolu Gbenga wrote: Good Day all Please I am having some issues on my voip asterisk server I make internal calls on extensions configured ie extension 192 can call extension 195 etc But each time i try to make calls outside the extension ie calling a GSM or an external line ,i always hear this response all trunk calls are busy please try your call again later Please how can i resolve this problem . I will appreciate your response. Best Regards Success You need to at least post some verbose from the console and explain how you are connecting to the PSTN. It would greatly help if you included the relevant portions of your extensions.conf. Thanks, Steve Totaro ___ --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] All trunk are busy please try your call again later
Hi all, I am grateful for our contribution so far . I followed dave advise and i have the attached file using the aterisk -r when a call is made. I attached two files. One of the attached file is for the external call,which replied with the PROBLEM all trunks are busy now,please try your call again later. The second attachment is when i made internal calls and the phone rang. Please,i will be expecting your replies for further directions. Best Regards On Dec 20, 2007 2:58 PM, Steve Totaro [EMAIL PROTECTED] wrote: What is the output of ztconfig from the Linux command line? What does your zaptel.conf and zapata.conf look like? What is the relevant part of extensions.conf (the dialout section that fails). Also from the CLI, it would be most helpful to post the output you get when dialing out fails. I don't think it is a network issue at all, I think your configs need some work. Thanks, Steve Totaro Lolu Gbenga wrote: Good Day Find attached the relevant portions of the asterisk CLI. Please,which portion of the extension .conf should i send ? It is connected via RJ 45 connector to an E1 modem to the telco company. I use E1 link. I will appreciate your reply. Best Regards On Dec 18, 2007 4:02 PM, dave cantera [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: lolu, sounds more like a telco/itsp problem then *. I would tcpdump -i eth0 port 5060 to make sure it is actually going out... change 5060 if you have changed your port to your itsp, of course. to see what is going on as well as the other debugging notes mentioned in this thread. daveC Lolu Gbenga wrote: Good Day all Please I am having some issues on my voip asterisk server I make internal calls on extensions configured ie extension 192 can call extension 195 etc But each time i try to make calls outside the extension ie calling a GSM or an external line ,i always hear this response all trunk calls are busy please try your call again later Please how can i resolve this problem . I will appreciate your response. Best Regards Success ___ --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 -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --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] All trunk are busy please try your call again later
7871 7871) in new stack -- Executing Set(SIP/7871-bb64, FROMCONTEXT=exten-vm) in new stack -- Executing Macro(SIP/7871-bb64, record-enable|7874|IN) in new stack -- Executing GotoIf(SIP/7871-bb64, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI(SIP/7871-bb64, recordingcheck|20051006-002614|1128554774. 10) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20051006-002614|1128554774.10: Inbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(SIP/7871-bb64, No recording needed) in new stack -- Executing Macro(SIP/7871-bb64, dial|15|tr|7874) in new stack -- Executing AGI(SIP/7871-bb64, dialparties.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi -- dialparties.agi: priority = 1 -- dialparties.agi: callingani2 = 0 -- dialparties.agi: accountcode = -- dialparties.agi: channel = SIP/7871-bb64 -- dialparties.agi: callerid = 7871 -- dialparties.agi: context = macro-dial -- dialparties.agi: callington = 0 -- dialparties.agi: dnid = 7874 -- dialparties.agi: request = dialparties.agi -- dialparties.agi: calleridname = 7871 -- dialparties.agi: extension = s -- dialparties.agi: language = en -- dialparties.agi: uniqueid = 1128554774.10 -- dialparties.agi: callingpres = 0 -- dialparties.agi: type = SIP -- dialparties.agi: rdnis = unknown -- dialparties.agi: callingtns = 0 -- dialparties.agi: enhanced = 0.0 dialparties.agi: Caller ID name is '7871' number is '7871' dialparties.agi: Methodology of ring is 'none' -- dialparties.agi: Added extension 7874 to extension map -- dialparties.agi: Extension 7874 cf is disabled -- dialparties.agi: Extension 7874 do not disturb is disabled -- dialparties.agi: Checking CW and CFB status for extension 7874 == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged on from 127.0.0.1 -- dialparties.agi: Correct AMPMGRUSER and AMPMGRPASS == Manager 'admin' logged off from 127.0.0.1 dialparties.agi: Extension 7874 is available...skipping checks -- dialparties.agi: DbSet CALLTRACE/7874 to 7871 -- AGI Script dialparties.agi completed, returning 0 -- Executing Dial(SIP/7871-bb64, SIP/7874|15|tr) in new stack -- Called 7874 -- SIP/7874-5b48 is ringing == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/7871-bb64' in ma cro 'dial' == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/7871-bb64' in ma cro 'exten-vm' == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/7871-bb64' asterisk1*CLI THANKS SO MUCH I WILL BE EXPECTING YOUR REPLY. On Dec 20, 2007 5:09 PM, Lolu Gbenga [EMAIL PROTECTED] wrote: Hi all, I am grateful for our contribution so far . I followed dave advise and i have the attached file using the aterisk -r when a call is made. I attached two files. One of the attached file is for the external call,which replied with the PROBLEM all trunks are busy now,please try your call again later. The second attachment is when i made internal calls and the phone rang. Please,i will be expecting your replies for further directions. Best Regards On Dec 20, 2007 2:58 PM, Steve Totaro [EMAIL PROTECTED] wrote: What is the output of ztconfig from the Linux command line? What does your zaptel.conf and zapata.conf look like? What is the relevant part of extensions.conf (the dialout section that fails). Also from the CLI, it would be most helpful to post the output you get when dialing out fails. I don't think it is a network issue at all, I think your configs need some work. Thanks, Steve Totaro Lolu Gbenga wrote: Good Day Find attached the relevant portions of the asterisk CLI. Please,which portion of the extension .conf should i send ? It is connected via RJ 45 connector to an E1 modem to the telco company. I use E1 link. I will appreciate your reply. Best Regards On Dec 18, 2007 4:02 PM, dave cantera [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: lolu, sounds more like a telco/itsp problem then *. I would tcpdump -i eth0 port 5060 to make sure it is actually going out... change 5060 if you have changed your port to your itsp, of course. to see what is going on as well as the other debugging notes mentioned in this thread. daveC Lolu Gbenga wrote: Good Day all Please I am having some issues on my voip asterisk server I make internal calls on extensions configured ie extension 192 can call
[asterisk-users] All trunk are busy please try your call again later
Good Day all Please I am having some issues on my voip asterisk server I make internal calls on extensions configured ie extension 192 can call extension 195 etc But each time i try to make calls outside the extension ie calling a GSM or an external line ,i always hear this response all trunk calls are busy please try your call again later Please how can i resolve this problem . I will appreciate your response. Best Regards Success ___ --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
