May 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] configuring gateway for making calls
by Samira Mh 14 May '12

14 May '12

hi, thanks so much for great help but while dialing my umber using extensino:1000 that register on myfreeswitch server the error is as follow: 2012-05-13 20:23:55.098050 [NOTICE] switch_channel.c:669 New Channel sofia/internal/1000(a)192.168.1.100 [5c057400-2be8-4080-a401-25b5dc402dd0] 2012-05-13 20:23:55.106289 [INFO] mod_dialplan_xml.c:418 Processing 1000->00989191949637 in context default 2012-05-13 20:23:55.111170 [NOTICE] switch_ivr.c:1447 Transfer sofia/internal/1000(a)192.168.1.100 to enum[00989191949637@default] 2012-05-13 20:23:55.248259 [INFO] switch_core_state_machine.c:142 No Route, Aborting 2012-05-13 20:23:55.248259 [NOTICE] switch_core_state_machine.c:143 Hangup sofia/internal/1000(a)192.168.1.100 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2012-05-13 20:23:55.257505 [NOTICE] switch_core_session.c:1182 Session 59 (sofia/internal/1000(a)192.168.1.100) Ended 2012-05-13 20:23:55.257505 [NOTICE] switch_core_session.c:1184 Close Channel sofia/internal/1000(a)192.168.1.100 [CS_DESTROY] 1: --------------------------------------------------------- cat /usr/local/freeswitch/conf/dialplan/default/01_shatel.xml <include> <context name="default"> <extension name="Dial Out Custom Gateway"> <condition field="destination_number" expression="^00(98/d{10})$"> <action application="bridge" data="sofia/gateway/shatel/$1"/> </condition> </extension> </context> </include> 2:cat /usr/local/freeswitch/conf/directory/default/1000.xml -------------------------------------------------- <include> <user id="1000"> <params> <param name="password" value="$${default_password}"/> <param name="vm-password" value="1000"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1000"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 1000"/> <variable name="effective_caller_id_number" value="1000"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include> ________________________________ From: Erkan Ünlü (simalube) <erkan(a)speedingtrade.com> To: 'Samira Mh' <saami_mh(a)ymail.com> Sent: Monday, May 14, 2012 3:09 PM Subject: AW: [Freeswitch-users] configuring gateway for making calls Hi Samira, you need 3 xml files. 1. Xml file is located at the directory/default Here put this xml sample and try after reloadxml to register your softphone <include> <user id="1000"> <params> <param name="password" value="yourpass" /> <param name="vm-password" value="1000" /> </params> <variables> <variable name="toll_allow" value="domestic,international,local" /> <variable name="accountcode" value="1000" /> <variable name="user_context" value="myWorldWideDialingPlan" /> <variable name="effective_caller_id_name" value="0000" /> <variable name="effective_caller_id_number" value="0000" /> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}" /> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}" /> <variable name="callgroup" value="techsupport" /> </variables> </user> </include> 2. Build your dialingplan The dialingplan is located in the dialingplan folder. Create here a new xml and put this sample inside the file. In the future you can study the default.xml in dialingplan folder to learn more about the dialingplan. In this section you see that the user_context in the directory folder XML file is the same with context_name So the user account is in association with the dialingplan. If you need you can build more different dialingplans for other requirements. <include> <context name="myWorldWideDialingPlan"> <extension name="Worldwide"> <condition field="destination_number" expression="^00 (\d+)$"> <action application="set" data="hangup_after_bridge=true" /> <action application="bridge" data="sofia/gateway/gateway/00$1|sofia/gateway/gateway2/00$1" /> </condition> </extension> </context> 3. The last step is the gateway Your gateway looks good and is reged correctly You see the gateway name you written “custom” You can change it to gateway1 or myfavoriteGW or what you want. But the name of the gatewayname must be the same like this line in section 2 of my description <action application="bridge" data="sofia/gateway/gateway/00$1|sofia/gateway/gateway2/00$1" /> Change the red gateway to the right gatewayname in your sip profiles external folder located XML You see in my sample "sofia/gateway/gateway/00$1|sofia/gateway/gateway2/00$1” I have 2 gateways for failover if one gateway is down or gives errors. | ßthese character is an OR and mean if the first gateway gives an error so try the second gateway You can use also only 1 gateway so delete the second gateway until to “ character Hope that is now clear for you. Kind regards Erkan Von:Samira Mh [mailto:saami_mh@ymail.com] Gesendet: Montag, 14. Mai 2012 12:26 An: FreeSWITCH Users Help Cc: erkan(a)speedingtrade.com Betreff: Re: [Freeswitch-users] configuring gateway for making calls hi erkan; thanks so much for your reply; i think your mean is: i must configure file 01_custom.xml like : <include> <extension name="Dial Out Custom Gateway"> <context name="samira"> <condition field="destination_number" expression="^9(1\d{10})$"> <action application="bridge" data="sofia/gateway/custom/$1"/> </condition> </context> </extension> </include> and because i registered via extension:1000 on freeswitch so : must configure file 1000.xml like : <include> <user id="1000"> <params> <param name="password" value="yourpass" /> <param name="vm-password" value="1000" /> </params> <variables> <variable name="toll_allow" value="domestic,international,local" /> <variable name="accountcode" value="1000" /> <variable name="user_context" value="linktoyourdialplan" /> <variable name="effective_caller_id_name" value="0000" /> <variable name="effective_caller_id_number" value="0000" /> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}" /> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}" /> <variable name="callgroup" value="techsupport" /> </variables> </user> </include> but i on't know what to be set as <variable name="user_context" value="linktoyourdialplan" /> in the 1000.xml? please help, i am new in freeswitch sorry for my english ;; ________________________________ From:Erkan Ünlü (simalube) <erkan(a)speedingtrade.com> To: 'FreeSWITCH Users Help' <freeswitch-users(a)lists.freeswitch.org> Sent: Monday, May 14, 2012 2:01 PM Subject: Re: [Freeswitch-users] configuring gateway for making calls Hi Samira, please build first of all them in the directory/default folder a XML file like this. <include> <user id="1000"> <params> <param name="password" value="yourpass" /> <param name="vm-password" value="1000" /> </params> <variables> <variable name="toll_allow" value="domestic,international,local" /> <variable name="accountcode" value="1000" /> <variable name="user_context" value="linktoyourdialplan" /> <variable name="effective_caller_id_name" value="0000" /> <variable name="effective_caller_id_number" value="0000" /> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}" /> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}" /> <variable name="callgroup" value="techsupport" /> </variables> </user> </include> Now you can register with your softphone to FS. See the user_context area in this section you must link it to your dialing plan The dialing plan is located in dialing plan folder. My simple dialing plan looks like this. <include> <context name="theNameOfDialingplan"> <extension name="Worldwide"> <condition field="destination_number" expression="^00 (\d+)$"> <action application="set" data="hangup_after_bridge=true" /> <action application="bridge" data="sofia/gateway/gateway/00$1|sofia/gateway/gateway2/00$1" /> </condition> </extension> </context> The context name of the dialing plan must be the same in the directory XML user_context I color it in red. I hope that this info are usefully for you. Kind regards. Von:freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von Samira Mh Gesendet: Montag, 14. Mai 2012 11:11 An: Free SWITCH Users Help Betreff: [Freeswitch-users] configuring gateway for making calls hi guys; i have configured gateway for making calls but it dosn't work; my config files are as follow: first create file that name " iptel.org.xml " in the below path: /usr/local/freeswitch/conf/sip_profiles/external/ i have an account on iptel.org like :(username:aimas;password:mypass) <include> <gateway name="custom"> <param name="username" value="arimas"/> <param name="password" value="mypass"/> <param name="realm" value="iptel.org"/>  <param name="proxy" value="sip.iptel.org"/> </gateway>--> </include> the run : sofia profile external restart reloadxml reloxm and sofia status: Name Type Data State ================================================================================================= external profile sip:mod_sofia@10.0.3.15:5080 RUNNING (0) external::example.com gateway sip:joeuser@example.com NOREG external::custom gateway sip:arimas@sip.iptel.org REGED internal profile sip:mod_sofia@10.0.3.15:5060 RUNNING (0) 10.0.3.15 alias internal ALIASED ================================================================================================= then create file name "01_custom.xml " in the below path : /usr/local/freeswitch/conf/dialplan/default the content of the files 01_custom.xml are: <include> <extension name="Dial Out Custom Gateway"> <condition field="destination_number" expression="^9(1\d{10})$"> <action application="bridge" data="sofia/gateway/custom/$1"/> </condition> </extension> </include> . now when i dial 9, 1-800-555-1212 on eyebeam that register on freeswitch server the call is failed please help; thanks _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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disable SIP privacy
by Michal Zubáč 14 May '12

14 May '12

Hi. I want to disable Privacy on outgoing SIP leg but I can't do it because incoming SIP leg has Privacy enabled. I want to set outgoing caller_id to some constant number for these cases. Can this even be done? I tried sip_Privacy variables, but with no effect. Thanks. Michal Zubac

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Phone not registering
by Miha 14 May '12

14 May '12

Hi, here is pastebin of siptrace (http://pastebin.freeswitch.org/19029) Phone on local network are registered on FS. After I put between local network and Phone router, phones are unable to registered on FS. On other softswitch which is not FS phones are registering (same port, same scenario, etc.). Phones are SPA922. What could be causing the problem? Thanks! Miha

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load average is hight (Very urgent please)
by ghallab 14 May '12

14 May '12

Hi all, This days am experiencing a problem with Freeswitch: it seems that it needs more hardware resources and voicemail don't work any more and it make a long delay to process call transfer. should I compile the last version to revolve the problem or try to increase hardware resources? I notice that I have anther instance of FS running fine in the same conditions. I tried to collect the maximum information that I can: OS version: CentOS release 5.4 the cpu info is: http://pastebin.freeswitch.org/19039 the mem info is: http://pastebin.freeswitch.org/19040 the out put of the command top is http://pastebin.freeswitch.org/19038 the number of calls when I made the test was: 5 total. the out put of gcore and gdb: http://pastebin.freeswitch.org/19041 number of sofia's profiles: 5 sofia profiles and 1 alias number of calls that I want: 30 simultaneous calls I want also to notice that I get all the time some warning messages like 2012-05-11 14:36:10.716722 [WARNING] sofia_reg.c:1442 SIP auth challenge (REGISTER) on sofia profile 'internal' for [7351@domain] from ip @ip

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error/NO_ROUTE_DESTINATION while define groupserror/NO_ROUTE_DESTINATION
by Samira Mh 14 May '12

14 May '12

Hello, i have defined the group named: "custom" in the below path as follow : /usr/src/freeswitch-1.0.6/conf/directory/default.xml <group name="custom"> <users> <user id="1005" type="pointer"/> <user id="1006" type="pointer"/> </users> </group> then issue "reloadxml" and run the comand: group_call custom but i come accross with the following error: error/NO_ROUTE_DESTINATION please help,all configs are defined correctly , but i couldn't define the "custom" group on freeswitch

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RTP PROXY MEDIA
by Andrew Paul 14 May '12

14 May '12

Hai , I have connected asterisk with freeswitch. In freeswitch i enabled the rtp-proxy-media equals true . Whenever i am making call from asterisk to freeswitch and in freeswitch side i put on hold am able to hear MOH from asterisk. But after that when i try any transfers (REFER) it wont coming to asterisk and it is still processiong in freeswitch only. In freeswithch any method to rely sip messages directly to other end . Thanks And Regards Andrew Paul

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Re: [Freeswitch-users] RTP media Issue
by Mr Nathan Downes 12 May '12

12 May '12

Hi Anthony, Thanks for the input, I am happy to leave it on if there is no performance penalty, it's a quad core xeon so it isn't even scratching the surface yet. The weird thing is if I do a wireshark on the interface that goes from endpoint - fs and one on the fs - trunk at the same time I don't see the same issue coming in, that is what was confusing me. Maybe the endpoint isn't handling the timestamp issue as well as FS is, because the problem they report is worse than what I hear in the trace. We can consider this issue resolved unless I find out it is causing some other issue, I haven't found jitter too much problem yet as it is all over fibre and I have a 2-3ms end to end latency. Keep up the good work on a brilliant product! From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Sunday, 13 May 2012 4:31 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] RTP media Issue With the option off the original timestamps from the inbound leg to Fs are passed out the outbound leg. With it on Fs creates new timestamps, there is no performance penalty but the disadvantage is that any jitter will he masked since the timestamps are changed . If the calls sound ok then its ok to leave it. On May 12, 2012 12:49 AM, "Mr Nathan Downes" <nathandownes(a)hotmail.com> wrote: Thanks for the response, Keys: Help Display mode Restart statistics Order of fields quit Packets Pings Host Loss% Snt Last Avg Best Wrst StDev 1. 10.77.0.254 0.0% 18911 0.2 0.2 0.2 0.2 0.0 2. 192.168.5.1 0.0% 18911 2.4 <tel:2.4%C2%A0%C2%A0%202.4%C2%A0%C2%A0%202.3%C2%A0%C2%A0%202.4%C2%A0%C2%A0%2 00.0> 2.4 2.3 2.4 0.0 3. 10.77.4.22 0.0% 18911 2.7 3.1 2.4 4.1 0.5 1 is my switch at DC, 2 is unknown on fibre, 3 is endpoint ATA device at site about 140km away Duplex is set correctly on all interfaces, I don't see anything on the fibre part of network it is all L2. I am not sure if it is a network issue, as the wireshark capture I was doing on the server that FS resides on, and it was reporting the stream going from FS to endpoint had the wrong timestamps. From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Robert-IPhone Sent: Saturday, 12 May 2012 10:09 AM To: FreeSWITCH Users Help Cc: <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] RTP media Issue Start with running an extended "mtr" between the locations. Maybe you are dealing with a latency issue. I've also seen issues with duplexing. So look at low level network issues Sent from my iPhone 4S On May 11, 2012, at 5:58 PM, Mr Nathan Downes <nathandownes(a)hotmail.com> wrote: Hi First time trying this mailing list, I have my FS server in a data centre, which feeds site a, site b etc. over a fibre network. We are having really bad call quality issues at one of the sites. The residents report that every second or 2 there is a hiccup in the incoming voice and they miss a portion of the voice. It is consistent and affects everybody. I did some wireshark tracing of both sides of the calls (ext - fs and fs - provider) and to me it appears packets coming from my server have a wrong timestamp every second or so. The server is not over loaded and is pretty default Ubuntu 10.04 LTS, latest FS Head on there. I spent a week trying to diagnose with both the fibre provider and trunk provider but we couldn't figure out the cause. I then tried putting rtp-rewrite-timestamps in both the profiles (external one to provider, and internal one specifically for fibre company) and the problem appears to have gone away. I would like to figure out what is/was causing the issue and also find out if that option puts any extra overhead on the server. If there is almost no overhead I would probably leave it on and just assume I need I for this specific connection. I don't have the issue at another site over same fibre network and trunk provider, and people connecting direct over internet don't suffer same problem. I have about 2gb of wireshark traces I can put online if required. Thanks, Nathan _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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max-db-handles
by Yuriy Nasida 12 May '12

12 May '12

Hello guys. I have question about max-db-handles.Is it for the simultaneous connections to core DB (sqlite) only ?I use lua scripts + mysql via freeswitch.Dbh from dialplan. Does max-db-handles will affects on mysql as well ? Please advise.Thanks.

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Re: [Freeswitch-users] RTP media Issue
by Mr Nathan Downes 12 May '12

12 May '12

Thanks for the response, Keys: Help Display mode Restart statistics Order of fields quit Packets Pings Host Loss% Snt Last Avg Best Wrst StDev 1. 10.77.0.254 0.0% 18911 0.2 0.2 0.2 0.2 0.0 2. 192.168.5.1 0.0% 18911 2.4 2.4 2.3 2.4 0.0 3. 10.77.4.22 0.0% 18911 2.7 3.1 2.4 4.1 0.5 1 is my switch at DC, 2 is unknown on fibre, 3 is endpoint ATA device at site about 140km away Duplex is set correctly on all interfaces, I don’t see anything on the fibre part of network it is all L2. I am not sure if it is a network issue, as the wireshark capture I was doing on the server that FS resides on, and it was reporting the stream going from FS to endpoint had the wrong timestamps. From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Robert-IPhone Sent: Saturday, 12 May 2012 10:09 AM To: FreeSWITCH Users Help Cc: <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] RTP media Issue Start with running an extended "mtr" between the locations. Maybe you are dealing with a latency issue. I've also seen issues with duplexing. So look at low level network issues Sent from my iPhone 4S On May 11, 2012, at 5:58 PM, Mr Nathan Downes <nathandownes(a)hotmail.com> wrote: Hi First time trying this mailing list, I have my FS server in a data centre, which feeds site a, site b etc. over a fibre network. We are having really bad call quality issues at one of the sites. The residents report that every second or 2 there is a hiccup in the incoming voice and they miss a portion of the voice. It is consistent and affects everybody. I did some wireshark tracing of both sides of the calls (ext – fs and fs – provider) and to me it appears packets coming from my server have a wrong timestamp every second or so. The server is not over loaded and is pretty default Ubuntu 10.04 LTS, latest FS Head on there. I spent a week trying to diagnose with both the fibre provider and trunk provider but we couldn’t figure out the cause. I then tried putting rtp-rewrite-timestamps in both the profiles (external one to provider, and internal one specifically for fibre company) and the problem appears to have gone away. I would like to figure out what is/was causing the issue and also find out if that option puts any extra overhead on the server. If there is almost no overhead I would probably leave it on and just assume I need I for this specific connection. I don’t have the issue at another site over same fibre network and trunk provider, and people connecting direct over internet don’t suffer same problem. I have about 2gb of wireshark traces I can put online if required. Thanks, Nathan _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Re: [Freeswitch-users] freeswitch and voipstunt
by Fernando Di Nucci 12 May '12

12 May '12

It is a routing issue. I have to fix my router-firewall. It lacks some iptables helper. Thank you all for your help

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