March 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

IVR to hunt group to external destination fails // direct to hunt group to external desitnation succeeds
by William Traenkle 15 Mar '10

15 Mar '10

When a hunt group is accessed from the main IVR, the hunt group dials an internal extension via transfer correctly but when trying to dial an external number such a cell phone via sip uri it is unable to connect or ring the phone but it waits the correct timeout period before it goes to voicemail. However, if you access the hunt group directly without going through the main IVR, all works perfectly. Any ideas where I should look and what line of code I might be missing? Thanks, -Will

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Fwd: Set/Export channel variable
by Ahmed Bhaila 15 Mar '10

15 Mar '10

I am looking at the FS Console and I have a ESL listener socket that is hooked for HANGUP_COMPLETE event. - ahmed On Mon, Mar 15, 2010 at 6:50 AM, Ahmed Ajmal <ahmed.ajmal at breezecom.ae <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>>wrote: >* Hi *>* *>* *>* *>* I have now configured the gateways as suggested below but I am not able to *>* see the sip_gateway_name variable when the call hangup event is complete. I *>* am on Freeswitch 1.0.5. *>* *>* *>* *Where are you looking to find the variable? -MC From: freeswitch-users-bounces at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> [mailto:freeswitch-users-bounces at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>] On Behalf Of Brian West Sent: Wednesday, March 10, 2010 7:08 PM To: freeswitch-users at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> Subject: Re: [Freeswitch-users] Set/Export channel variable We already set you a variable called sip_gateway_name when a call is placed out a gateway. You're also doing this wrong you should NEVER EVER do @$ipgwX as that data is in the gateway. You should simply do sofia/gateway/gwname/number /b On Mar 10, 2010, at 8:01 AM, Ahmed Ajmal wrote: Hi I am trying to set a channel variable on the bridge application using this: <action application="bridge" data="[gw=gateway1]sofia/gateway/gw/$number@$ipgw1 | [gw=gateway2]sofia/gateway/gw/$number@$ipgw2"/> The application runs fine, what I now need to do is set/export the 'gw' channel variable for CDR so that I know which gateway was dialed. I have tried using set and export application but that doesn't work. Please help. Thanks Ahmed

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FreeSWITCH Conf Call Agenda
by Michael Collins 15 Mar '10

15 Mar '10

Hello all, The agenda for Wednesday's conference call is now up: http://wiki.freeswitch.org/wiki/FS_weekly_2010_03_17 We could use some more content. Also, we would love to have a volunteer discuss one of the FreeSWITCH modules or some other FS feature that he/she uses. If you have any thoughts or suggestions please email me off list. Thanks! -MC

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100% CPU
by Spencer Thomason 15 Mar '10

15 Mar '10

Hi, I have Freeswitch running as a SBC in front of several Asterisk machines and without warning one of the Freeswitch processes will use 100% of one of the CPU cores and stop responding to SIP requests. I'm using the latest svn on Centos 5.4 in a xen vm. Basically everything will be fine for about 3-4 hours and then this happens. [root@sip ~]# cat /proc/cpuinfo processor : 0 vendor_id : AuthenticAMD cpu family : 15 model : 33 model name : AMD Opteron(tm) Processor 275 stepping : 2 cpu MHz : 2204.998 cache size : 1024 KB physical id : 0 siblings : 1 core id : 0 cpu cores : 1 fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu de tsc msr pae cx8 apic cmov pat clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt lm 3dnowext 3dnow pni cmp_legacy bogomips : 5514.25 TLB size : 1024 4K pages clflush size : 64 cache_alignment : 64 address sizes : 40 bits physical, 48 bits virtual power management: ts fid vid ttp processor : 1 vendor_id : AuthenticAMD cpu family : 15 model : 33 model name : AMD Opteron(tm) Processor 275 stepping : 2 cpu MHz : 2204.998 cache size : 1024 KB physical id : 1 siblings : 1 core id : 0 cpu cores : 1 fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu de tsc msr pae cx8 apic cmov pat clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt lm 3dnowext 3dnow pni cmp_legacy bogomips : 5514.25 TLB size : 1024 4K pages clflush size : 64 cache_alignment : 64 address sizes : 40 bits physical, 48 bits virtual power management: ts fid vid ttp processor : 2 vendor_id : AuthenticAMD cpu family : 15 model : 33 model name : AMD Opteron(tm) Processor 275 stepping : 2 cpu MHz : 2204.998 cache size : 1024 KB physical id : 2 siblings : 1 core id : 0 cpu cores : 1 fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu de tsc msr pae cx8 apic cmov pat clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt lm 3dnowext 3dnow pni cmp_legacy bogomips : 5514.25 TLB size : 1024 4K pages clflush size : 64 cache_alignment : 64 address sizes : 40 bits physical, 48 bits virtual power management: ts fid vid ttp processor : 3 vendor_id : AuthenticAMD cpu family : 15 model : 33 model name : AMD Opteron(tm) Processor 275 stepping : 2 cpu MHz : 2204.998 cache size : 1024 KB physical id : 3 siblings : 1 core id : 0 cpu cores : 1 fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu de tsc msr pae cx8 apic cmov pat clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt lm 3dnowext 3dnow pni cmp_legacy bogomips : 5514.25 TLB size : 1024 4K pages clflush size : 64 cache_alignment : 64 address sizes : 40 bits physical, 48 bits virtual power management: ts fid vid ttp [root@sip ~]# uname -r 2.6.18-164.11.1.el5xen -- Spencer Thomason 5Nine Solutions LLC e. sales(a)5ninesolutions.com p. +1.888.271.7959 f. +1.310.510.6980

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Dial number at registered extension
by Spencer Thomason 15 Mar '10

15 Mar '10

Hello all, Is there a way I can dial at number at a registered extension. Basically what I'd like to do is use Freeswitch as a SBC for a number of asterisk machines and I'd like to pass an incoming sip invite to a specific number at each one. Right now I'm specifying the IP in the bridge command like this sofia/default/NNNNNNNNNNN(a)xx.xx.xx.xx but if I dial the registered user directly it dials the contact address at the registered user and does not pass the inbound number. Basically what I'm trying to do is this: pstn -> ITSP -> freeswitch -> asterisk (1-n) Thanks, Spencer

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FreeSWITCH to add SRTP to Asterisk
by schogge 15 Mar '10

15 Mar '10

Hi Community, I'm new to FreeSWITCH. What I have read so far it seems to be a very interesting project. For our company we are running an Asterisk PBX in a datacenter. After some years it is doing everything we want except of SRTP. Would it be possible to run FreeSWITCH in front of Asterisk to convert SRTP coming from our phones to RTP which is ok for Asterisk and vice versa? Thanks, schogge -- View this message in context: http://old.nabble.com/FreeSWITCH-to-add-SRTP-to-Asterisk-tp27901964p2790196… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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problem in sending calls to internal Gateway.
by Prabin Shrestha 15 Mar '10

15 Mar '10

hello, My extension are working properly. But they can't send calls to gateway. I have 2 interfaces, one with public and another with private lan. Gateway is in private lan. Please suggest. My configurations are as follows: /conf/sip_profiles/external.xml: <param name="rtp-ip" value="110.4.12.111"/> <param name="sip-ip" value="110.4.12.111"/> <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> /conf/sip_profiles/internal.xml: <param name="rtp-ip" value="192.168.1.11"/> <param name="sip-ip" value="192.168.1.11"/> <param name="hold-music" value="$${hold_music}"/> <param name="apply-nat-acl" value="nat.auto"/> haven't made any changes in conf/autoload_configs/acl.conf.xml to route calls from providers, /dialplan/public/providers.xml: <extension name="provider"> <condition field="network_addr" expression="200.144.44.177"/> <condition field="destination_number" expression="^(\d+)$"> <action application="transfer" data="$1 XML context_AX"/> </condition> </extension> gateway is defined at /conf/sip_profiles/internal/gateways.xml: <include> <gateway name="gateway_AX"> <param name="username" value="admin"/> <param name="password" value="nepal123"/> <param name="proxy" value="192.168.1.100"/> <param name="register" value="false"/> <param name="extension" value="1001"/> </gateway> </include> To route internal calls from extensions, /conf/dialplan/default.xml: <extension name="my_internal">  <condition field="destination_number" expression="^(d+)$"> <action application="bridge" data="sofia/gateway/gateway_AX/$1"/> </condition> </extension> to route calls from providers, conf/dialplan/inbound_routing.xml: <context name="context_AX"> <extension name="from_provider_to_AX">  <condition field="destination_number" expression="^9(\d+)$"> <action application="bridge" data="sofia/gateway/gateway_AX/$1"/> </condition> </extension> </context> Please help. The INTERNET now has a personality. YOURS! See your Yahoo! Homepage. http://in.yahoo.com/

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not answering calls on TDM400 port
by Brian May 15 Mar '10

15 Mar '10

Hello, I noticed Freeswitch wasn't answering calls on one of the TDM400 ports yesterday. All other ports seemed to be OK. Nothing showed up on the console, set at debug log level. I restarted Freeswitch and it came good. What should I do if it happens again? Is there any debugging information I can extract from the system? If so how? Thanks -- Brian May <brian(a)microcomaustralia.com.au>

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Re: [Freeswitch-users] Set/Export channel variable
by Brian West 15 Mar '10

15 Mar '10

We already set you a variable called sip_gateway_name when a call is placed out a gateway. You're also doing this wrong you should NEVER EVER do @$ipgwX as that data is in the gateway. You should simply do sofia/gateway/gwname/number /b On Mar 10, 2010, at 8:01 AM, Ahmed Ajmal wrote: > Hi > > I am trying to set a channel variable on the bridge application using this: > > <action application="bridge" data="[gw=gateway1]sofia/gateway/gw/$number@$ipgw1 | [gw=gateway2]sofia/gateway/gw/$number@$ipgw2"/> > > The application runs fine, what I now need to do is set/export the ‘gw’ channel variable for CDR so that I know which gateway was dialed. I have tried using set and export application but that doesn’t work. Please help. > > Thanks > Ahmed

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errors and failures at snagoma+freeswitch
by Tzury Bar Yochay 15 Mar '10

15 Mar '10

Hi, Owe to an hardware failure I have had to reinstall my server. Originally I have had the combination of FreeSWITCH/Sangoma/LibPRI/OpenZap Getting recommendation to try the new sangoma prid I tried to follow the instructions found at http://wiki.sangoma.com/wanpipe-freeswitch-install Being unable to make it work, I search through the mailing-list and saw the specific versions which should be used. Followed these instruction as well, yet, it is not working. probing hardware shows all goos indications. When I try to originate from within the FreeSWITCH's CLI I get freeswitch@snaptrunk-02> originate openzap/smg_prid/a/0525133399 XML public 2010-03-14 16:09:14.575286 [CRIT] ozmod_sangoma_boost.c:254 SPAN is not online. -ERR NORMAL_CIRCUIT_CONGESTION 2010-03-14 16:09:14.575286 [ERR] switch_ivr_originate.c:2422 Cannot create outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION] freeswitch@snaptrunk-02> when I try to run #smg_ctrl start I get: smg_ctrl: sangoma_prid failed to start > smg_ctrl: check /var/log/sangoma_mgd.log for errors > *However, that file ( /var/log/sangoma_mgd.log) is empty. *I pastebin all relevant files at http://gist.github.com/332589* *I am using the latest freeswitch (svn trunk) I tried the smg with these versions (from ftp://ftp.sangoma.com/linux/custom/DavidYS/) - wanpipe-3.5.10.smg_pri.1.tgz - wanpipe-3.5.10.3.smg_pri-v1.58.tgz * *

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