June 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Forwarding SIP calls between 2 VoIP carriers without using SIP TRUNKING
by mosbah abdelkader 28 Jun '11

28 Jun '11

Hi, One client have 2 SIP accounts: SIP1 and SIP2 with 2 different carriers: C1 and C2. He wants to forward all calls destined to SIP1 to SIP2. C2 is using FreeSWITCH. I don't want to register SIP1 as SIP trunk in C2 because if the number of SIP accounts is big, there will be a performance problem as each account will be registered as a SIP trunk. Can anyone help me by giving an idea and a config sample for FreeSWITCH. Thank you. -- Best Regards

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Phrase macros in conferences
by Bryan Smart 28 Jun '11

28 Jun '11

I'm attempting to use phrase macros in mod_conference, but I'm not having success. Perhaps they aren't supported? I have a working phrase macro. In the dialplan, both <action application="phrase" data="my_phrase"/> and <action application="playback" data="phrase:my_phrase"/> Work just fine. The wiki says that, in conference.conf.xml, entries like pin-sound accept, for their value, file paths, strings prefixed with "say" for TTS, and strings prefixed with "tone_stream". This looks almost the same as what can be passed to the playback application in the dialplan. When I try to prefix a string with "phrase", it doesn't work. <param name="pin-sound" value="phrase:my_phrase"/> The console tells me "unknown file format". The console still tells me this if I make up phrase names <param name="pin-sound" value="phrase:not_a_real_phrase"/> So it is complaining about the syntax or capability, rather than an unrecognized format of an actual file. If mod_conference doesn't support phrase macros, is this a near-future feature? Phrases would seem to be necessary to ocalize announcements and play announcements that are stitched together from multiple files. Bryan

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xml_curl directory - reject registration with specific error code?
by Dmitry Sytchev 28 Jun '11

28 Jun '11

Hi all! Is there a way to reject user with specific error code / message when i use XML directory via curl? For example, I'd like to distinguish between user blocked / user not found errors and so on. In other words, instead of returning default directory stub and 404, I want FS to return some specific code on failed registration attempt. Thanks! -- Best regards, Dmitry Sytchev, IT Engineer

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Hooks for own call logs
by Sascha Daniels 28 Jun '11

28 Jun '11

Hi together. My dialplan is generated by a java script. I am writing call logs (not cdr) in the application that feeds the js. Everething is fine except canceled calls. I think I need a hook in the dialplan. How can I make a system call (or what ever) when a call is canceled? Any hints? Regards Sascga

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SIP/2.0 403 Forbidden
by Al Bogner 28 Jun '11

28 Jun '11

I am doing my first steps with freeswitch. I have installed on an Ubuntu 11.04 server: ii freeswitch 1.0.7~20110603-0natty6 ii freeswitch-codec-passthru-g7231 1.0.7~20110603-0natty6 ii freeswitch-codec-passthru-g729 1.0.7~20110603-0natty6 ii freeswitch-lang-de 1.0.7~20110603-0natty6 ii freeswitch-lang-en 1.0.7~20110603-0natty6 Following the tutorial at http://www.onlinesolution.co.nz/viewtopic.php?t=102 I modified /opt/freeswitch/conf/directory/default/1000.xml <include> <user id="1000"> <params> <param name="password" value="siptest"/> <param name="vm-password" value="siptest"/> </params> <variables> <variable name="toll_allow" value="domestic,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> Then I tried to connect to the server from another machine: +++ 27-6-2011 17:07:30.279796 INFO SIP ::send_sip_udp Send to: udp:192.168.2.100:5060 REGISTER sip:192.168.2.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.108;rport;branch=z9hG4bKcvkohckk Max-Forwards: 70 To: "FS Admin" <sip:1000@192.168.2.100> From: "FS Admin" <sip:1000@192.168.2.100>;tag=nyyvz Call-ID: zmjxppampyrrgxk(a)client.local.tld CSeq: 645 REGISTER Contact: <sip:1000@192.168.2.108>;expires=3600 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE User-Agent: Twinkle/1.4.2 Content-Length: 0 --- +++ 27-6-2011 17:07:30.281799 INFO SIP ::process_sip_msg Received from: udp:192.168.2.100:5060 SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.2.108;received=192.168.2.108;rport=5060;branch=z9hG4bKcvkohckk To: "FS Admin" <sip:1000@192.168.2.100>;tag=iairy From: "FS Admin" <sip:1000@192.168.2.100>;tag=nyyvz Call-ID: zmjxppampyrrgxk(a)client.local.tld CSeq: 645 REGISTER Server: Twinkle/1.4.2 Content-Length: 0 What could be wrong, so I can't connect? Al

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does FreeSWITCH care about beeing connected to an IPv4- or IPv6-ISP ?
by bill evergreen 28 Jun '11

28 Jun '11

Hello, I intend to setup a little FreeSWITCH test-environment. Does FreeSWITCH care about beeing connected to an IPv4- OR IPv6-isp (it should be possible to operate the same environment under IPv4 OR IPv6) ? Thank's a lot for any feedback! Bill PS I am new to FreeSWITCH :-)

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Reject SIP registrations
by Kurtis Heimerl 28 Jun '11

28 Jun '11

Hello FS Users! I'm trying to create the following setup. When a user registers, if they register on a known account (lets say X), they do not need a password. X's registration is immediately OK'd, and everything is great. I've gotten that working using the ACL. The IP address of our SIP clients are added through cidr and the clients do not need to give passwords. However, for some reason, if another account that does not exist in the directory (let's say Y) registers, FS returns with a 200 OK, instead of rejecting Y. I'm trying to figure out why this is the case, and how to remedy that fact. I have the following line in my internal.xml file, which I had assumed would force this function:  <param name="inbound-reg-force-matching-username" value="true"/> However, it does not work. In my directory, each individual account as the following lines: <user id="1303"> <params> <param name="accept-blind-reg" value="true"/> <param name="accept-blind-auth" value="true"/> <param name="vm-password" value="1000"/> </params> Though I've found that removing it (from all users in the directory) doesn't help. I'm primarily concerned with the line in internal.xml; it seems possible that the fact that we do not have an auth-user (because we do not require auth) means that this won't work. However, I have yet to test that hypothesis. The ACL has been the most confusing aspect of this installation, with a lot of undocumented aspects, and I get the nagging feeling this is another. I could very well be wrong though. Thanks for any direction.

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.wav vs .gsm file sizes for recording calls.
by Wes 28 Jun '11

28 Jun '11

In my tests, if I record a call in .wav format, a 10 second file is about 177,000 bytes, while a 10 second .gsm file is 17,000 bytes. I then used sox to convert the .gsm file to a .wav file, and it stayed at around 17,000 bytes. So, is the default recording format for .wav using a higher sample rate? vs the default conversion format for the sox tool? checking the file type using "file" I see that the larger one is: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz and the wav created by sox via the default conversion from .gsm is: RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz So apparently the larger wav file is 16 bit... how are these recording parameters controlled? Can I set it to record directly into the smaller wav format? Or will I have to run sox on every file...

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Re: [Freeswitch-users] Reject SIP registrations
by Peter Olsson 28 Jun '11

28 Jun '11

AFAIK this is the intended behaviour. If allowed through ACL using cidr, the user is treated as authenticated, so no registration is required. /Peter ----- Reply message ----- Från: "Kurtis Heimerl" <kheimerl(a)cs.berkeley.edu> Datum: tis, jun 28, 2011 01:19 Rubrik: [Freeswitch-users] Reject SIP registrations Till: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> One of those links got screwed up... Anyhow, here are those three config files: internal.xml : http://pastebin.freeswitch.org/16609 acl.conf.xml : http://pastebin.freeswitch.org/16610 1300.xml : http://pastebin.freeswitch.org/16611 If anything else could help, I'd love to share it. The basic story, so far as I see, is that I allow specific IPs through the ACL. Somehow this is allowing ANY SIP username to register, rather than just those defined (such as 1300). Any help would be appreciated. On Mon, Jun 27, 2011 at 4:11 PM, Kurtis Heimerl <kheimerl(a)cs.berkeley.edu> wrote: > Anyhow, here are those three config files: > > internal.xml : http://pastebin.freeswitch.org/16609 > acl.conf.xml : http://pastebin.freeswitch.org/16610 > 1300.xml : http://pastebin.freeswitch.org/16611 > > If anything else could help, I'd love to share it. > > The basic story, so far as I see, is that I allow specific IPs through > the ACL. Somehow this is allowing ANY SIP username to register, rather > than just those defined (such as 1300). Any help would be appreciated. > > On Mon, Jun 27, 2011 at 1:30 PM, Kurtis Heimerl > <kheimerl(a)cs.berkeley.edu> wrote: >> It's enabled in the acl.conf.xml file, using CIDR. >> >> What conf files do you consider relevant? acl.conf.xml, internal.xml, >> a profile or two, anything else? >> >> On Mon, Jun 27, 2011 at 1:26 PM, David Ponzone <david.ponzone(a)ipeva.fr> wrote: >>> The interesting question is then: why are you able to register without >>> password, if this feature is not enabled on the profile... >>> Perhaps you should recap your config once more, and put the relevant files >>> on PB. >>> David Ponzone Direction Technique >>> email: david.ponzone(a)ipeva.fr >>> tel: 01 74 03 18 97 >>> gsm: 06 66 98 76 34 >>> Service Client IPeva >>> tel: 0811 46 26 26 >>> www.ipeva.fr<http://www.ipeva.fr> - www.ipeva-studio.com<http://www.ipeva-studio.com> >>> Ce message et toutes les pièces jointes sont confidentiels et établis à >>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion >>> non autorisée est interdite. Tout message électronique est susceptible >>> d'altération. IPeva décline toute responsabilité au titre de ce message s'il >>> a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce >>> message, merci de le détruire immédiatement et d'avertir l'expéditeur. >>> >>> >>> >>> Le 27/06/2011 à 20:36, Kurtis Heimerl a écrit : >>> >>> That would explain why removing them didn't do anything! >>> >>> Thanks. >>> >>> On Mon, Jun 27, 2011 at 6:25 AM, Steven Ayre <steveayre(a)gmail.com> wrote: >>> >>> Just so you know... >>> >>> <param name="accept-blind-reg" value="true"/> >>> >>> <param name="accept-blind-auth" value="true"/> >>> >>> These will have no effect in the user directory. They only apply to SIP >>> >>> profiles. >>> >>> -Steve >>> >>> >>> >>> On 27 June 2011 02:23, Kurtis Heimerl <kheimerl(a)cs.berkeley.edu> wrote: >>> >>> Hello FS Users! >>> >>> I'm trying to create the following setup. When a user registers, if >>> >>> they register on a known account (lets say X), they do not need a >>> >>> password. X's registration is immediately OK'd, and everything is >>> >>> great. I've gotten that working using the ACL. The IP address of our >>> >>> SIP clients are added through cidr and the clients do not need to give >>> >>> passwords. >>> >>> However, for some reason, if another account that does not exist in >>> >>> the directory (let's say Y) registers, FS returns with a 200 OK, >>> >>> instead of rejecting Y. I'm trying to figure out why this is the case, >>> >>> and how to remedy that fact. >>> >>> I have the following line in my internal.xml file, which I had assumed >>> >>> would force this function: >>> >>>  >>> >>> <param name="inbound-reg-force-matching-username" value="true"/> >>> >>> However, it does not work. In my directory, each individual account as >>> >>> the following lines: >>> >>> <user id="1303"> >>> >>> <params> >>> >>> <param name="accept-blind-reg" value="true"/> >>> >>> <param name="accept-blind-auth" value="true"/> >>> >>> <param name="vm-password" value="1000"/> >>> >>> </params> >>> >>> Though I've found that removing it (from all users in the directory) >>> >>> doesn't help. >>> >>> I'm primarily concerned with the line in internal.xml; it seems >>> >>> possible that the fact that we do not have an auth-user (because we do >>> >>> not require auth) means that this won't work. However, I have yet to >>> >>> test that hypothesis. The ACL has been the most confusing aspect of >>> >>> this installation, with a lot of undocumented aspects, and I get the >>> >>> nagging feeling this is another. I could very well be wrong though. >>> >>> Thanks for any direction. >>> >>> _______________________________________________ >>> >>> Join us at ClueCon 2011, Aug 9-11, Chicago >>> >>> http://www.cluecon.com 877-7-4ACLUE >>> >>> 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 >>> >>> >>> _______________________________________________ >>> >>> Join us at ClueCon 2011, Aug 9-11, Chicago >>> >>> http://www.cluecon.com 877-7-4ACLUE >>> >>> 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 >>> >>> >>> >>> _______________________________________________ >>> Join us at ClueCon 2011, Aug 9-11, Chicago >>> http://www.cluecon.com 877-7-4ACLUE >>> >>> 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 >>> >>> >>> _______________________________________________ >>> Join us at ClueCon 2011, Aug 9-11, Chicago >>> http://www.cluecon.com 877-7-4ACLUE >>> >>> 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 >>> >>> >> > _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE 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 !DSPAM:4e090ed032762115621071!

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building rpm doesn't get me sounds directory
by Dave Horton 28 Jun '11

28 Jun '11

I'm building an rpm using the freeswitch.spec in the top-level directory, and it creates a usable rpm for me except that when I install it on a target machine, I don't get the sounds sub-directory and therefore voicemail doesn't work. Do I need to make some changes to the spec file or the rpm build script in order to get an rpm that will install all of the sounds/recordings?

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