Re: [asterisk-users] speciality of SIPp and SER(Sip Express Router)
Dear Faisal Hanif, Thank you for you reply. I got my point. Thanks in advance Nahar On Tue, Aug 10, 2010 at 3:52 PM, Faisal Hanif wrote: > Hi, > > SER is a carrier grade SIP Server/Proxy and used in large scale SIP > networks like Verizon. It can do lot of functionality SIP registration, call > routing, load-balancing. Normally i is used for clustering of billing > servers. > > SIPp is a software which can generate dummy voice calls to test any VoIP > platform. > > Regards, > > Faisal Hanif > *VoIP Manager > ***Vopium A/S ** > > On 8/10/2010 11:33 AM, kamrun nahar bina wrote: > > Dear Faisal Hanif, > > Thanks for your reply. > What is the purpose of using SER ? > What is the purpose of using SIPp -I know little bit about this. > > But I know nothing about SER? Could you please explain it ? Or in which > case it is necessary to use SER. > Please let me know? > > Thanks in advance > > Nahar > > On Tue, Aug 10, 2010 at 2:54 PM, Faisal Hanif wrote: > >> Hi, >> >> SER is a most powerful SIP router but a SIPp is a VoIP load generation >> software. So both are totally different and can not be used interchangably. >> >> Regards, >> >> Faisal Hanif >> *VoIP Manager >> ***Vopium A/S** >> >> On 8/10/2010 10:44 AM, kamrun nahar bina wrote: >> >> Dear all, >> >> What is the difference between SIPp and SER(Sip Express Router)? Which >> one is better load performance testing? >> Is there any one who knows about this? Could you please give me details >> informtaion? >> >> Thans in advance >> >> Nahar >> >> >> -- >> _ >> -- 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] speciality of SIPp and SER(Sip Express Router)
Dear Faisal Hanif, Thanks for your reply. What is the purpose of using SER ? What is the purpose of using SIPp -I know little bit about this. But I know nothing about SER? Could you please explain it ? Or in which case it is necessary to use SER. Please let me know? Thanks in advance Nahar On Tue, Aug 10, 2010 at 2:54 PM, Faisal Hanif wrote: > Hi, > > SER is a most powerful SIP router but a SIPp is a VoIP load generation > software. So both are totally different and can not be used interchangably. > > Regards, > > Faisal Hanif > *VoIP Manager > ***Vopium A/S** > > On 8/10/2010 10:44 AM, kamrun nahar bina wrote: > > Dear all, > > What is the difference between SIPp and SER(Sip Express Router)? Which one > is better load performance testing? > Is there any one who knows about this? Could you please give me details > informtaion? > > Thans in advance > > Nahar > > > -- > _ > -- 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] speciality of SIPp and SER(Sip Express Router)
Dear all, What is the difference between SIPp and SER(Sip Express Router)? Which one is better load performance testing? Is there any one who knows about this? Could you please give me details informtaion? Thans in advance Nahar -- _ -- 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] Want to retrieve the value of contact header
Dear Jim Dickenson. Ｔｈａｎｋｓ for you mail. I have got the solution. Thanks Nahar On Thu, Jul 1, 2010 at 11:45 AM, Jim Dickenson wrote: > You might take a look at the SIPHEADER function which can return specific > SIP headers. > -- > Jim Dickenson > mailto:dicken...@cfmc.com > > CfMC > http://www.cfmc.com/ > > > > On Jun 30, 2010, at 7:36 PM, kamrun nahar bina wrote: > > Dear all, > > I want to retrieve the value from Contact header and from "From header " > which is "0345001280" from the following two lines: > Contact: > > From: "99 " > > >;tag=as191896a1 > > Is it possible in asterisk to retrieve that value? I am getting following > value in the corresponding variable when I pass the following SIP message. > Is there anything which contain the value of "0345001280" of contact ? > Corresponding value: > CALLERID(num): 185475 > CALLERID(name) : 99 > SCI-PEERNAME : 185475 > > SIP message: > > INVITE sip:08058913...@113.34.235.106 > SIP/2.0 > Via: SIP/2.0/UDP 123.50.217.143:5060;branch=z9hG4bK100b063a;rport > From: "99 " > > >;tag=as191896a1 > To: > > Contact: > > Call-ID: 0f3fbfe3463035d04f05534824a18...@113.34.235.106 > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Thu, 01 Jul 2010 02:20:18 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Type: application/sdp > Content-Length: 267 > > v=0 > o=root 22702 22702 IN IP4 123.50.217.143 > s=session > c=IN IP4 123.50.217.143 > t=0 0 > m=audio 17262 RTP/AVP 0 8 3 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > > > Is it possible to retrieve the value of contact in asterisk ? Please let me > know. > Is there anyone who knows the solution? I need this urgent. > > Thanks in advance > > Nahar > -- > _ > -- 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
[asterisk-users] Want to retrieve the value of contact header
Dear all, I want to retrieve the value from Contact header and from "From header " which is "0345001280" from the following two lines: Contact: > From: "99 " >;tag=as191896a1 Is it possible in asterisk to retrieve that value? I am getting following value in the corresponding variable when I pass the following SIP message. Is there anything which contain the value of "0345001280" of contact ? Corresponding value: CALLERID(num): 185475 CALLERID(name) : 99 SCI-PEERNAME : 185475 SIP message: INVITE sip:08058913...@113.34.235.106 SIP/2.0 Via: SIP/2.0/UDP 123.50.217.143:5060;branch=z9hG4bK100b063a;rport From: "99 " >;tag=as191896a1 To: > Contact: > Call-ID: 0f3fbfe3463035d04f05534824a18...@113.34.235.106 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 01 Jul 2010 02:20:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 267 v=0 o=root 22702 22702 IN IP4 123.50.217.143 s=session c=IN IP4 123.50.217.143 t=0 0 m=audio 17262 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - Is it possible to retrieve the value of contact in asterisk ? Please let me know. Is there anyone who knows the solution? I need this urgent. Thanks in advance Nahar -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Channel cannot be released
Dear all, using asterisk-1.4.23.1, I encountered a problem of asterisk that cannot release the channel.* * We have several of asterisk server(client ,Guest). Now channels remaining problem occurs only in the server where the number of user agent is more than 660 and where many simultaneous calling occurs. Physically, it is being released, but in programming logic, it is not being released. If we execute "core show channels concise" then we see that the channels is remaining in server which is not using long time. Is it the bugs of asterisk or something else? if asterisk has limitation then how many concurrent call can occur in asterisk? Or how many user agent can register in one asterisk server? Or is it the server load problem? Or is it the problem of configuration file settings? We have specified the value of canreinvite is "no" . Please let me know. We have got the channels remaining problem in the following hand set. Acrobits Softphone version 3.2.2 (iPhone) SipSimple v4.0/iPhoneOS snom300/7.1.30 Grandstream HT487 1.0.8.16 Linphone/Linphone-3.1.2 (eXosip2/unknown)...for fax Sipdroid(Linksys/PAP2-3.1.22( LS) Is there any one who knows the solution? Please help me. Thanks in advance Nahar -- _ -- 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] problem of "Cannot release Channel"
Dear all, using asterisk-1.4.23.1, I encountered a problem of asterisk that cannot release the channel.* * We have several of asterisk server(client ,Guest). Now channels remaining problem occurs only in the server where the number of user agent is more than 660 and where many simultaneous calling occurs. Physically, it is being released, but in programming logic, it is not being released. If we execute "core show channels concise" then we see that the channels is remaining in server which is not using long time. Is it the bugs of asterisk or something else? if asterisk has limitation then how many concurrent call can occur in asterisk? Or how many user agent can register in one asterisk server? Or is it the server load problem? Please let me know. We have got the channels remaining problem in the following hand set. Acrobits Softphone version 3.2.2 (iPhone) SipSimple v4.0/iPhoneOS snom300/7.1.30 Grandstream HT487 1.0.8.16 Linphone/Linphone-3.1.2 (eXosip2/unknown)...for fax Sipdroid(Linksys/PAP2-3.1.22(LS) Is there any one who knows the solution? Please help me. Thanks in advance Nahar -- _ -- 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] Problem of "Playing 'pbx-transfer'"
Our codec is ulaw. We tested snom to snom, x-lite to x-lite. We are getting same problems as usual. I alse tested for another device like linksys to snom, x-lite to snom. mobile phone to snom or x-lite. The same problem occurs for above two options. We sometimes hear transfer' s sound, sometime cannot hear transfer sound. The same problem for answering machine. Is it the load problem or something else? Is there any solution for this problem? our system is like as: The number of User agent is: 1650 The number of Actual registered user agent is: 600 version of asterisk: 1.4.23.1 our memory size is 4GB. concurrent calls no : 30. Thanks in advance Nahar On Mon, May 10, 2010 at 8:25 PM, Adolphe Cher-aime wrote: > Try x-lite to x-lite, snom to snom . That may be a codec problem. > > Which codec are you using? > > > Adolphe Cher-aime > From my Iphone > > On May 9, 2010, at 11:11 PM, "Dovid Bender" > wrote: > > Process of elemination. Test with multiple phones, check the codec being > used and make sure the file is there and available. > > > - Original Message - > *From:* kamrun nahar bina > *To:* asterisk-users@lists.digium.com > *Sent:* Friday, May 07, 2010 07:33 > *Subject:* [asterisk-users] Problem of "Playing 'pbx-transfer'" > > Dear all, > > We have been using asterisk for 4 years. Now we have got problems which > occurs during the attended transfer. > During attended transfer, sometimes we cannot hear the sound of > 'pbx-transfer'. > > I cannot understand why this is happening? > log is : > > > -- Started music on hold, class 'default', on SIP/113.34.235.13-b7a3f110 > > -- Playing 'pbx-transfer' (language 'jp') > > > Although it is showing Playing 'pbx-transfer' (language 'jp'), but it cannot > hear 'pbx-transfer' sound > Sometimes we can hear the sound of 'pbx-transfer'. > is it the problem of network load or phone-set or something else? Please let > me know. I am using x-lite and snom 300. > > Before i tested it for memory load, And found out that it is not a memory > problem. > > Our system is as like as: > The number of User agent is: 1650 > The number of Actual registered user agent is: 600 > > > Our System configuration is : > > IBM X3550 > CPU: Intel(R) Xeon(R) CPU X5460 @ 3.16GHz > > HDD: 3.5 SATA 1TB x 2 > version of asterisk: 1.4.23.1 > > our memory size is 4GB. > concurrent calls no : 30. > Our memory condition is below : > > > > Cpu(s): 0.3%us, 0.7%sy, 0.0%ni, 98.5%id, 0.0%wa, 0.1%hi, 0.3%si, > > 0.0%st > Mem: 4147888k total, 3986540k used, 161348k free,76852k buffers > Swap: 2031608k total, 56k used, 2031552k free, 3170396k cached > > > PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND > > 23160 root 15 0 440m 415m 5688 S 4.3 10.3 398:13.93 asterisk > > Our disk space condition is below: > FilesystemSize Used Avail Use% Mounted on > > /dev/mapper/VolGroup00-LogVol00 > 901G 245G 610G 29% / > > /dev/sda1 99M 18M 77M 19% /boot > tmpfs 2.0G 0 2.0G 0% /dev/shm > > > Asterisk and the User-Agent is connected through the Internet. > > ..And Is there any solution to solve this problem? I have > investigated in several places but I cannot find out the reason? > I need this solution very urgently. Is there any one who can solve this > problem? > > -- > _ > -- 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> > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: ><http://lists.digium.com/mailman/listinfo/asterisk-users> > 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] Problem of "Playing 'pbx-transfer'"
Dear, We have tested in several phone like snom 300, LINKSYS, s-lite. The same error occurs in every phone. Our codec is ulaw. and we have checked that the file is there and available. Is there any other solution? or is it the problem of db load, is it possible? Thanks in advance. nahar On Mon, May 10, 2010 at 1:11 PM, Dovid Bender wrote: > Process of elemination. Test with multiple phones, check the codec being > used and make sure the file is there and available. > > > - Original Message - > *From:* kamrun nahar bina > *To:* asterisk-users@lists.digium.com > *Sent:* Friday, May 07, 2010 07:33 > *Subject:* [asterisk-users] Problem of "Playing 'pbx-transfer'" > > Dear all, > > We have been using asterisk for 4 years. Now we have got problems which > occurs during the attended transfer. > During attended transfer, sometimes we cannot hear the sound of > 'pbx-transfer'. > > I cannot understand why this is happening? > log is : > > > -- Started music on hold, class 'default', on SIP/113.34.235.13-b7a3f110 > > -- Playing 'pbx-transfer' (language 'jp') > > > Although it is showing Playing 'pbx-transfer' (language 'jp'), but it cannot > hear 'pbx-transfer' sound > Sometimes we can hear the sound of 'pbx-transfer'. > is it the problem of network load or phone-set or something else? Please let > me know. I am using x-lite and snom 300. > > Before i tested it for memory load, And found out that it is not a memory > problem. > > Our system is as like as: > The number of User agent is: 1650 > The number of Actual registered user agent is: 600 > > > Our System configuration is : > > IBM X3550 > CPU: Intel(R) Xeon(R) CPU X5460 @ 3.16GHz > > HDD: 3.5 SATA 1TB x 2 > version of asterisk: 1.4.23.1 > > our memory size is 4GB. > concurrent calls no : 30. > Our memory condition is below : > > > > Cpu(s): 0.3%us, 0.7%sy, 0.0%ni, 98.5%id, 0.0%wa, 0.1%hi, 0.3%si, > > 0.0%st > Mem: 4147888k total, 3986540k used, 161348k free,76852k buffers > Swap: 2031608k total, 56k used, 2031552k free, 3170396k cached > > > PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND > > 23160 root 15 0 440m 415m 5688 S 4.3 10.3 398:13.93 asterisk > > Our disk space condition is below: > FilesystemSize Used Avail Use% Mounted on > > /dev/mapper/VolGroup00-LogVol00 > 901G 245G 610G 29% / > > /dev/sda1 99M 18M 77M 19% /boot > tmpfs 2.0G 0 2.0G 0% /dev/shm > > > Asterisk and the User-Agent is connected through the Internet. > > ..And Is there any solution to solve this problem? I have > investigated in several places but I cannot find out the reason? > I need this solution very urgently. Is there any one who can solve this > problem? > > > -- > _ > -- 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] Problem of hearing attended transfer' s sound
Dear all, We have been using asterisk for 4 years. Now we have got problems which occurs during the attended transfer. During attended transfer, sometimes we cannot hear the sound of 'pbx-transfer'. Sometimes we can hear little portion of 'pbx-transfer's sound. That means sound also become noisy. I cannot understand why this is happening? log is : -- Started music on hold, class 'default', on SIP/113.34.235.13-b7a3f110 -- Playing 'pbx-transfer' (language 'jp') Although it is showing Playing 'pbx-transfer' (language 'jp'), but we cannot hear 'pbx-transfer' sound Sometimes we can hear the sound of 'pbx-transfer' and sometimes we cannot hear. is it the problem of server load or memory load or something else? Please let me khow. The message of asterisk log is always same.so it is very difficult to understand the problem. Our system is as like as: The number of User agent is: 1650 The number of Actual registered user agent is: 600 Asterisk and the User-Agent is connected through the Internet. ..And Is there any solution to solve this problem? We have investigated in several places but we cannot find out the reason? We need this solution very urgently. Is there any one who can solve this problem? Thanks in advance Nahar -- _ -- 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] Problem of hearing transfer' s sound
Dear all, We have been using asterisk for 4 years. Now we have got problems which occurs during the attended transfer, blind transfer and answering machine(during the pressing of 999). During attended transfer and blind transfer , sometimes we cannot hear the sound of 'pbx-transfer'. The same problem occurs during answering machine. I cannot understand why this is happening? log is : -- Started music on hold, class 'default', on SIP/113.34.235.13-b7a3f110 -- Playing 'pbx-transfer' (language 'jp') Although it is showing Playing 'pbx-transfer' (language 'jp'), but we cannot hear 'pbx-transfer' sound Sometimes we can hear the sound of 'pbx-transfer' and sometimes we cannot hear. is it the problem of server load or memory load or something else? Please let me khow. The message of asterisk log is always same.so it is very difficult to understand the problem. Our system is as like as: The number of User agent is: 1650 The number of Actual registered user agent is: 600 Asterisk and the User-Agent is connected through the Internet. ..And Is there any solution to solve this problem? We have investigated in several places but we cannot find out the reason? We need this solution very urgently. Is there any one who can solve this problem? Please help me. Thanks in advance Nahar -- _ -- 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] Problem of "Playing 'pbx-transfer'"
Dear all, We have been using asterisk for 4 years. Now we have got problems which occurs during the attended transfer. During attended transfer, sometimes we cannot hear the sound of 'pbx-transfer'. I cannot understand why this is happening? log is : -- Started music on hold, class 'default', on SIP/113.34.235.13-b7a3f110 -- Playing 'pbx-transfer' (language 'jp') Although it is showing Playing 'pbx-transfer' (language 'jp'), but it cannot hear 'pbx-transfer' sound Sometimes we can hear the sound of 'pbx-transfer'. is it the problem of network load or phone-set or something else? Please let me know. I am using x-lite and snom 300. Before i tested it for memory load, And found out that it is not a memory problem. Our system is as like as: The number of User agent is: 1650 The number of Actual registered user agent is: 600 Our System configuration is : IBM X3550 CPU: Intel(R) Xeon(R) CPU X5460 @ 3.16GHz HDD: 3.5 SATA 1TB x 2 version of asterisk: 1.4.23.1 our memory size is 4GB. concurrent calls no : 30. Our memory condition is below : Cpu(s): 0.3%us, 0.7%sy, 0.0%ni, 98.5%id, 0.0%wa, 0.1%hi, 0.3%si, 0.0%st Mem: 4147888k total, 3986540k used, 161348k free,76852k buffers Swap: 2031608k total, 56k used, 2031552k free, 3170396k cached PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 23160 root 15 0 440m 415m 5688 S 4.3 10.3 398:13.93 asterisk Our disk space condition is below: FilesystemSize Used Avail Use% Mounted on /dev/mapper/VolGroup00-LogVol00 901G 245G 610G 29% / /dev/sda1 99M 18M 77M 19% /boot tmpfs 2.0G 0 2.0G 0% /dev/shm Asterisk and the User-Agent is connected through the Internet. ..And Is there any solution to solve this problem? I have investigated in several places but I cannot find out the reason? I need this solution very urgently. Is there any one who can solve this problem? -- _ -- 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] problem of "when memory become 50% or more then sound become noisy?"
Dear Tzafrir Cohen, Now I executed "vmstat 1", Now memory usage is 15% thats why (swap in) and so (swap out) is 0. But When memory usage become 50% or more then swap size become 224172 kB according to previous log. May be this is the reason for becoming sound noisy? But How i will solve this memory's problem of asterisk? Thanks in advance Nahar On Tue, Apr 13, 2010 at 7:04 PM, Tzafrir Cohen wrote: > On Tue, Apr 13, 2010 at 06:42:50PM +0900, kamrun nahar bina wrote: > > Dear all, > > > > Currently I am using asterisk 1.4.23.1. . Over the period of 1 week, > > the memory in use starts off at 50% > > Is there much active swapping? > > Run 'vmstat 1' for a while. > > Look at the columns 'si' (swap in) and so (swap out). > > -- > 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
[asterisk-users] problem of "when memory become 50% or more then sound become noisy?"
Dear all,
Currently I am using asterisk 1.4.23.1. . Over the period of 1 week,
the memory in use starts off at 50% and
continues to climb until it hits 99%. When memory usage ratio become
50% or more, the quality of calls become
extremely noisy. The call quality goes back to being perfect once I
reboot the machine,
but I was to try and avoid having to reboot the machine every week.
the following is the memory status during the usage ratio of memory
approx. 50% or more which is in /proc/{process id of asterisk}/smaps
file :
09001000-7ebf6000 rw-p 09001000 00:00 0 [heap]
Size: 1929172 kB
Rss:1679436 kB
Shared_Clean: 0 kB
Shared_Dirty: 0 kB
Private_Clean: 6472 kB
Private_Dirty: 1672964 kB
Swap: 224172 kB
My server's RAM size is:
MemTotal: 4147888 kB
Processor is :
Intel(R) Xeon(R) CPU X5460 @ 3.16GHz
Asterisk and the User-Agent is connected through the Internet.
Is it the prablem of memory leakage of asterisk? Is there any solution
to solve this memory's problem?
is it asterisk's bug or something else? I cannot find out the solution
and cannot find out where is the problem?
Presently, I need this solution very urgently. I am eagerly waiting for reply.
Or is there any solution to clean up the memory's usage space in
asterisk source code?
Thanks in advance
Nahar
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Re: [asterisk-users] "Failed to play transfer sound! " during attended transfer
Dear sir, Thanks for your reply. We have tested in another phone like Bria(2.4.3 buid 50906) with same phenomenon. But we are getting same error "Failed to play transfer sound! " during attended transfer. Is there anything which causes this problem? And we are not facing this problem first time. Before we faced in this problem occasionally. But recently, this problem occurs frequently. Is there any other problem or any other prerequisite for this problem? Or is it the problem of asterisk? How we can overcome this problem ? Please give us solution. Thanks in advance Nahar On Sat, Mar 27, 2010 at 1:33 AM, Alyed wrote: > so doesn't looks like overload > > Could it be a problem with the firmware of your softphones? Have you been > using some new phones lately? someone else in a different thread pointed on > attended transfer bugs with SNOM phones. > > > > We are eagerly waiting for your solution. > Hope we can help but don't so much pressure on me or the listers :) > > Alyed > > > > 2010/3/26 kamrun nahar bina > > Dear sir, >> >> Thanks for your reply. >> >> our memory size is 4GB. >> concurrent calls no : 30. >> Our memory condition is below : >> >> Cpu(s): 0.3%us, 0.7%sy, 0.0%ni, 98.5%id, 0.0%wa, 0.1%hi, 0.3%si, >> 0.0%st >> Mem: 4147888k total, 3986540k used, 161348k free,76852k buffers >> Swap: 2031608k total, 56k used, 2031552k free, 3170396k cached >> >> PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND >> 23160 root 15 0 440m 415m 5688 S 4.3 10.3 398:13.93 asterisk >> >> Our disk space condition is below: >> FilesystemSize Used Avail Use% Mounted on >> /dev/mapper/VolGroup00-LogVol00 >> 901G 245G 610G 29% / >> /dev/sda1 99M 18M 77M 19% /boot >> tmpfs 2.0G 0 2.0G 0% /dev/shm >> >> >> We are eagerly waiting for your solution. >> >> Thanks in advance. >> >> Nahar >> >> >> >> On Fri, Mar 26, 2010 at 2:32 PM, Alyed wrote: >> >>> If you didn't have this problem before I'll check up for any changes >>> lately (i suppose you have done so, but ask this just to be safe) >>> I see you have lots of agents and also lots of hard disk space, so I >>> guess disk space is not an issue. Please check it anyway. >>> >>> how many concurrent calls you have? 2 GB in RAM seems little against 600 >>> registered agents. >>> >>> Alyed >>> >>> >>> 2010/3/25 kamrun nahar bina >>> >>>> Dear sir, >>>> >>>> We have been using asterisk for 4 years. Now we have got problems which >>>> occurs during the attended transfer. >>>> But we are not always getting this problem. Sometimes it happens. But >>>> now we cannot understand why this is happening? >>>> >>>> problem is:"Failed to play transfer sound! " >>>> >>>> The log of asterisk is as like as followings: >>>> >>>> [Mar 25 17:58:40] DEBUG[23168] chan_sip.c: Invalid SIP message - >>>> rejected , no callid, len 366 >>>> >>>> >>>> >>>> >>>> [Mar 25 17:58:40] DEBUG[13617] audiohook.c: Write factory 0x1bfa000c was >>>> pretty quick last time, waiting for them. >>>> [Mar 25 17:58:40] DEBUG[13579] audiohook.c: Read factory 0x1bd61fe0 was >>>> pretty quick last time, waiting for them. >>>> >>>> >>>> >>>> >>>> [Mar 25 17:58:40] DEBUG[23168] chan_sip.c: Setting SIP_ALREADYGONE on >>>> dialog 5bd1acee539e699b4f5e79c94a348...@113.34.235.8 >>>> >>>> [Mar 25 17:58:40] DEBUG[23168] chan_sip.c: Received bye, issuing owner >>>> hangup >>>> [Mar 25 17:58:40] DEBUG[13617] audiohook.c: Write factory 0x1bfa000c was >>>> pretty quick last time, waiting for them. >>>> [Mar 25 17:58:40] WARNING[13591] res_features.c: Failed to play transfer >>>> >>>> >>>> >>>> >>>> sound! >>>> >>>> Our system is as like as: >>>> The number of User agent is: 1650 >>>> The number of Actual registered user agent is: 600 >>>> >>>> Our System configuration is : >>>> IBM X3550 >>>> CPU: Intel(R) Xeon(R) CPU X5460 @ 3.16GHz >>>> >>>> >>>> >>>> >>>> Memory: 2GB >>>> HDD
Re: [asterisk-users] "Failed to play transfer sound! " during attended transfer
Dear sir, Thanks for your reply. our memory size is 4GB. concurrent calls no : 30. Our memory condition is below : Cpu(s): 0.3%us, 0.7%sy, 0.0%ni, 98.5%id, 0.0%wa, 0.1%hi, 0.3%si, 0.0%st Mem: 4147888k total, 3986540k used, 161348k free,76852k buffers Swap: 2031608k total, 56k used, 2031552k free, 3170396k cached PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 23160 root 15 0 440m 415m 5688 S 4.3 10.3 398:13.93 asterisk Our disk space condition is below: FilesystemSize Used Avail Use% Mounted on /dev/mapper/VolGroup00-LogVol00 901G 245G 610G 29% / /dev/sda1 99M 18M 77M 19% /boot tmpfs 2.0G 0 2.0G 0% /dev/shm We are eagerly waiting for your solution. Thanks in advance. Nahar On Fri, Mar 26, 2010 at 2:32 PM, Alyed wrote: > If you didn't have this problem before I'll check up for any changes lately > (i suppose you have done so, but ask this just to be safe) > I see you have lots of agents and also lots of hard disk space, so I guess > disk space is not an issue. Please check it anyway. > > how many concurrent calls you have? 2 GB in RAM seems little against 600 > registered agents. > > Alyed > > > 2010/3/25 kamrun nahar bina > >> Dear sir, >> >> We have been using asterisk for 4 years. Now we have got problems which >> occurs during the attended transfer. >> But we are not always getting this problem. Sometimes it happens. But now >> we cannot understand why this is happening? >> >> problem is:"Failed to play transfer sound! " >> >> The log of asterisk is as like as followings: >> >> [Mar 25 17:58:40] DEBUG[23168] chan_sip.c: Invalid SIP message - >> rejected , no callid, len 366 >> >> >> [Mar 25 17:58:40] DEBUG[13617] audiohook.c: Write factory 0x1bfa000c was >> pretty quick last time, waiting for them. >> [Mar 25 17:58:40] DEBUG[13579] audiohook.c: Read factory 0x1bd61fe0 was >> pretty quick last time, waiting for them. >> >> >> [Mar 25 17:58:40] DEBUG[23168] chan_sip.c: Setting SIP_ALREADYGONE on >> dialog 5bd1acee539e699b4f5e79c94a348...@113.34.235.8 >> >> [Mar 25 17:58:40] DEBUG[23168] chan_sip.c: Received bye, issuing owner >> hangup >> [Mar 25 17:58:40] DEBUG[13617] audiohook.c: Write factory 0x1bfa000c was >> pretty quick last time, waiting for them. >> [Mar 25 17:58:40] WARNING[13591] res_features.c: Failed to play transfer >> >> >> sound! >> >> Our system is as like as: >> The number of User agent is: 1650 >> The number of Actual registered user agent is: 600 >> >> Our System configuration is : >> IBM X3550 >> CPU: Intel(R) Xeon(R) CPU X5460 @ 3.16GHz >> >> >> Memory: 2GB >> HDD: 3.5 SATA 1TB x 2 >> version of asterisk: 1.4.23.1 >> >> Asterisk and the User-Agent is connected through the Internet. >> >> >> ..And Is there any solution to solve this problem? We have investigated >> in several places but we cannot find out the reason? >> We need this solution very urgently. We are eagerly waiting for reply. >> >> Thanks in advance >> >> >> Nahar >> >> >> -- >> _ >> -- 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
[asterisk-users] "Failed to play transfer sound! " during attended transfer
Dear sir, We have been using asterisk for 4 years. Now we have got problems which occurs during the attended transfer. But we are not always getting this problem. Sometimes it happens. But now we cannot understand why this is happening? problem is:"Failed to play transfer sound! " The log of asterisk is as like as followings: [Mar 25 17:58:40] DEBUG[23168] chan_sip.c: Invalid SIP message - rejected , no callid, len 366 [Mar 25 17:58:40] DEBUG[13617] audiohook.c: Write factory 0x1bfa000c was pretty quick last time, waiting for them. [Mar 25 17:58:40] DEBUG[13579] audiohook.c: Read factory 0x1bd61fe0 was pretty quick last time, waiting for them. [Mar 25 17:58:40] DEBUG[23168] chan_sip.c: Setting SIP_ALREADYGONE on dialog 5bd1acee539e699b4f5e79c94a348...@113.34.235.8 [Mar 25 17:58:40] DEBUG[23168] chan_sip.c: Received bye, issuing owner hangup [Mar 25 17:58:40] DEBUG[13617] audiohook.c: Write factory 0x1bfa000c was pretty quick last time, waiting for them. [Mar 25 17:58:40] WARNING[13591] res_features.c: Failed to play transfer sound! Our system is as like as: The number of User agent is: 1650 The number of Actual registered user agent is: 600 Our System configuration is : IBM X3550 CPU: Intel(R) Xeon(R) CPU X5460 @ 3.16GHz Memory: 2GB HDD: 3.5 SATA 1TB x 2 version of asterisk: 1.4.23.1 Asterisk and the User-Agent is connected through the Internet. ..And Is there any solution to solve this problem? We have investigated in several places but we cannot find out the reason? We need this solution very urgently. We are eagerly waiting for reply. Thanks in advance Nahar -- _ -- 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
