- FreeSWITCH-users - FreeSWITCH Mailing Lists

Quick question on Originate through api
by Phillip Jones 21 May '10

21 May '10

Hi there, In my mind - where 6775557767 is a registered user, the following two are the same: apiResult = fsApi.Execute("originate", string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU',origination_caller_id_number=5556667776}}user/6775557767 ' apiResult = fsApi.Execute("originate", string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU',origination_caller_id_number=5556667776}}sofia/internal/6775557767(a)121.763.76.76' i.e. user/ and sofia/internal are the same thing. However on my system the user/ works great but sofia/internal loops call back through in the public context. Can someone tell me the difference between these options? What am I missing? Thanks Phil

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ISDN to SIP via Freeswitch - 408 Timeout and Terrible line noise
by samwise 21 May '10

21 May '10

Hi, I've just joined a team which had an Asterisk 1.6.0.1 desktop machine with a Digium TE122 card running CentOS 4.7 to bridge calls from, as far as I can tell, an E1 PRI line (with 10 DDI numbers) and software SIP UAs. Unfortunately, the machine died recently and will no longer boot from hard disk. So I've taken the card out and placed it in a HP Compaq ProLiant ML350 G3 Server with 1 GB RAM running CentOS 5.4 (Final). I've set it up to run the Freeswitch 1.0.6 release with the OpenZAP module enabled and pre-requisites dahdi-linux-complete-2.3.0+2.3.0 and libpri-1.4.10.2. All these components were built from source packages. I installed the default configuration files and configured everything as far as I could by following the instructions on the Freeswitch wiki. I've managed to get as far as successfully being able to dial one of the DDI numbers and getting the call to be delivered to an X-Lite SIP UA I have registered with Freeswitch. Unfortunately, I have two issues: 1) I have configured a SIP user account (Greg) but when I try to register X-Lite, it consistently returns a Registration error: 408 - Request Timeout. However, if I leave X-Lite alone, most of the time it will eventually (after quite some time) re-register with the server of it's own accord, allowing me to try a test call from the PSTN. 2) Whilst the SIP UA can then hear the call fine, there is a *lot* of noise on the PSTN end of the call - so much so that it's basically unusable, even though I can hear some things from the SIP end. Some info on the kernel/CPU included belom and some possibly useful logs / config files available here: openzap.conf: http://pastebin.freeswitch.org/13019 openzap.conf.xml: http://pastebin.freeswitch.org/13020 Greg.xml: http://pastebin.freeswitch.org/13021 61522x.xml: http://pastebin.freeswitch.org/13022 freeswitch.log: http://pastebin.freeswitch.org/13023 My question is where to start looking to diagnose these problems? I can provide whatever logs may be appropriate though I can't see any major problems in the freeswitch.log currently. With regards to the line noise issue, I read in the OpenZAP FAQ that unlike Asterisk, Freeswitch does not include a software echo canceller. Could that be related to the line noise? Should I attempt to install OSLEC to improve the quality? Thanks for any help - I'm fairly new to the world of telephony tecnologies like ISDN, Sam. # uname -a Linux bob 2.6.18-164.el5 #1 SMP Thu Sep 3 03:33:56 EDT 2009 i686 i686 i386 GNU/Linux # cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 6 model : 8 model name : Pentium III (Coppermine) stepping : 6 cpu MHz : 864.011 cache size : 256 KB fdiv_bug : no hlt_bug : no f00f_bug : no coma_bug : no fpu : yes fpu_exception : yes cpuid level : 2 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 mmx fxsr sse up bogomips : 1728.02

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kamailio and freeswitch integration
by Daniel-Constantin Mierla 21 May '10

21 May '10

Hello, I put together a tutorial about using kamailio (openser) and freeswtich together: the proxy takes care of authentication and registration, freeswitch of media services, here is a link: http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms Is it ok to upload it to FS wiki so others can add to it? Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

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control sip_cid_type from sip_profile
by Frank ＠ Impact 21 May '10

21 May '10

for RPID/PAI: sip_cid_type - Is there a way to control this setting in the gateway profile instead of having to do it in the bridge setup? Some providers seem to want rpid and others want p-asserted.

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Voip fail over device solutions, please help
by Carlos Lopez 21 May '10

21 May '10

Hi all, I am planning to deploy some test over a voip server, I'll be setting up two voip servers, but I want to know if there are devices that can take one T1/E1 PSTN lease and then transform that signal into IP and connect it to a ethernet card on each voip server. Also I'd like to know if I can do the same with the FXS analog phones which will be connected to a telphone patch panel and from that patch panel to a TDM 50pins PCI card, for example: 1- phone1<--->||Telephone ||<--Amphenolcable-->|Failover|<--->*1 phone2<--->||patch panel|| |device..|<--->*2 2- (*1-VOIP_SRV1)<--IP-->|Ethernet|<--TDMoE--->|Failover|<--->T1/E1 (*2-VOIP_SRV2)<--IP-->|Switch | |device | What are the best IP to analog phones PCI cards supported by Freeswitch, any hints?. Thanks in advanced for your help. Carlos.

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CDRs for Gateways
by Code Ghar 21 May '10

21 May '10

Using mod_cdr_csv, it is easy to create CDRs for registered users. However, how can we get CDRs for external gateways? For example, FreeSWITCH is configured to receive calls from one gateway and send them out through another. In this scenario, when neither inbound nor outbound gateway registers itself to FreeSWITCH, how can we generate and store CDRs for inbound gateway? Similarly, let's say a SIP provider sends a call meant for a registered user. For example, if a cell phone calls 6175550000, it is routed through the telephone network to the SIP provider. This provider routes the call to FreeSWITCH, which looks up in its dial plan that this number should be routed to extension (or registered user) 1000. There are no CDRs generated for either the SIP provider's gateway or the registered user in this case. In both these scenarios, if a call is inbound to FreeSWITCH but has not been generated by a registered user, mod_cdr_csv does not create CDRs. I have even tried to set the following parameter in cdr_csv.conf.xml but it didn't help. <param name="legs" value="ab"/> My question is this: can mod_cdr_csv generate CDRs for inbound calls from sources other than registered users? If it can't, how can this be achieved?

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Get Registered For ClueCon - You Could Win An iPad or MacBook Pro!
by Michael Collins 20 May '10

20 May '10

ClueCon is coming up fast! We are gearing up for the best ClueCon event yet. New sponsors, new speakers, and new prizes! This year we are giving away not one, but TWO Apple 16GB Wifi iPads! Additionally, there will be a 13" MacBook Pro to be given away. As we did last year, we will be laser engraving <http://cluecon.com/node/38> the ClueCon sponsors' logos on the back of the devices. If your organization wants to sponsor this year then contact us immediately so that you don't miss the deadline for having your logo included. ClueCon MMX is being held at the incredibly luxurious Trump Tower Hotel<http://cluecon.com/hotel-details>in downtown Chicago. Be sure to get registered right away - rooms are filling up quickly. You definitely don't want to miss this year's event. Call 877.742.CLUE to get registered. Remember, those who attended ClueCon 2009 qualify for a $150 discount off of the regular $699 conference price. That's right - those who attended last year can register again this year for only $549! Additionally, those who attend the FreeSWITCH training<http://freeswitchtraining.eventbrite.com/>in San Francisco, June 28-30, qualify for a special ClueCon rate of $500! Call today and take advantage of these great deals! Looking forward to seeing you in Chicago this August, The ClueCon Team http://www.cluecon.com 877.742.CLUE

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mod_dptools/redirect and 300 Multiple Choices
by Kristian Kielhofner 20 May '10

20 May '10

Hello everyone, I'd like to use redirect to send 300 Multiple Choices (with multiple contacts) instead of 302 Moved Temporarily with one contact. I tried using redirect and separating multiple contacts with spaces but FS just sent out a 500 :(. Is this possible? If not, how hard would it be to implement this feature? Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com

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nibblebill never hangs up
by Shrouk Khan 20 May '10

20 May '10

hi , i have successfully installed the nibble bill and trying to test billing depending on the destination , but even though the Cash goes to negetive the call never hangs up ! here are my conf : nibblebill.conf ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------ <configuration name="nibblebill.conf" description="Nibble Billing"> <settings>   <param name="db_username" value="root"/> <param name="db_password" value="fusionpbx"/> <param name="db_dsn" value="freeswitch_odbc"/>  <param name="db_table" value="v_extensions"/>  <param name="db_column_cash" value="Cash"/>  <param name="db_column_account" value="extension"/>    <param name="global_heartbeat" value="30"/>  <param name="lowbal_amt" value="5"/> <param name="lowbal_action" value="play ding"/>  <param name="nobal_amt" value="0.00"/> <param name="nobal_action" value="hangup"/>  <param name="percall_max_amt" value="100"/> <param name="percall_action" value="hangup"/> </settings> </configuration> ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------ an extension called 5400 <include> <user id="5400"> <params> <param name="password" value="5400"/> <param name="vm-password" value="8975"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="5400"/> <variable name="user_context" value="default"/> <variable name="nibble_account" value="5400"/> </variables> </user> </include> ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------ finally in the dial plan : <extension name="ToIcelandPBX.58d2" > <condition field="destination_number" expression="^(58.*)"> <action application="set" data="$${nibble_account}"/> </condition> <condition field="destination_number" expression="^(5807)$"> <action application="set" data="nibble_rate=30"/> <action application="bridge" data="sofia/gateway/ToIcelandPBX/$1"/> </condition> <condition field="destination_number" expression="^(5812)$"> <action application="set" data="nibble_rate=15"/> <action application="bridge" data="sofia/gateway/ToIcelandPBX/$1"/> </condition> </extension> right now the Cash balance for the extension is -192.00 and the extension can still make calls jsut fine. Am I missing something here ? -- Regards Shrouk Khan (Khan)

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Recommendations for adapter replacement ...
by Garrison Hoffman 20 May '10

20 May '10

I'm finally convinced my Grandstream HT386 is too much trouble. There's got to be some clever engineering in a device which manages to have a problem with every FS configuration I've tested. What I need now is another dual FXS adapter, one port is for a cordless phone and the other is a dedicated fax line. I'm considering an unlocked Linksys PAP2T, and a Grandstream HT502 but I'm open to other recommendations. The HT502 is only on my list due to UPnP & TLS support and because my GX2020 has always worked well. Both products are available at voipsupply.com for under $60, although I have no fixed price cutoff. -gh

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