May 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

phone hangs when placing internal calls
by Chris Brannon 28 May '13

28 May '13

Hello, I'm very new to freeswitch and telephony in general. I've managed to register phones with my server, but when I place test calls, the phone hangs, even though fs_cli tells me that the call was completed successfully. Also, everything looks fine when I run sip trace. FWIW, I've tried both linphone on Linux and csipsimple on Android. All of this is taking place on my LAN, so NAT isn't an issue. I started looking at traffic with tcpdump, and here's what I found. The phone authenticates to FreeSwitch and initiates the call. FreeSwitch sends a "100 trying" SIP message, followed by a UDP packet containing what appears to be SDP. It has no SIP header. This is followed by RTP from FreeSwitch to the phone. There is never a "200 OK" SIP message. Next, I ran linphone in debugging mode. I'll include a link to the log, in the hope that it will be helpful. Apparently, linphone isn't parsing the packet after the 100 trying message. Any idea what is going on here? http://the-brannons.com/linphone.log -- Chris

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voicemail is not working
by juned 28 May '13

28 May '13

Hi All, I am newbie to FS. so as a startup i have installed FS in mu local system to test out the basic functionality and features. so i have registered default users 1000 and 1001 in softphone( twinkle <http://mfnboer.home.xs4all.nl/twinkle/> ). registration was successful and calls was also fine but when i tried to check voicemail then it didn't worked. what i did to test it out voicemail is, did call to 1001 and let it rang so after 30 second if no answer is there then voicemail will be activated but calls are released after 30 seconds. As per documentation i came to know that by default voicemail is activated in default extensions is it so ? or i am missing something. Please point me to right direction, i want to have such a dialplan in which user can leave voicemail in cases of busy,unavailable and not answering. Thanks & Regards Juned -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/voicemail-is-not-working-tp75… Sent from the freeswitch-users mailing list archive at Nabble.com.

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ESL using bridge app doesn't return which gateway was used
by Clinton Goudie-Nice 28 May '13

28 May '13

When you make a bridge command using esl, where you specify multiple gateways or sip dials separated by or bars, you can't figure out which gateway was used. For example, if you bridge to something like this: sofia/gateway/SBC-GW2/+18019600000|sofia/gateway/SBC-GW1/+18019600000 The call could be bridged to either GW2 or GW1. When the CHANNEL_BRIDGE event is returned, you can see the original string in variable_current_application_data, and you may be able to infer the destination based on IP address, but nothing clearly says what gateway is used. If you turn on the all events firehose, you can see the CHANNEL_CREATE event come over the socket, and it does contain variable_sip_gateway_name with the actual name of the gateway, however I can't devise a way to access that data using the org.freeswitch.esl.client library, and even if I could, I still don't want all events for this system. Is it possible to get this information returned in any meaningful way through the ESL layer, either by an api command to query, or the setting of a variable that will give me back which gateway a bridge was performed through? If none of that is possible, this sounds worthy of filing a bug to return the variable_sip_gateway_name in the CHANNEL_BRIDGE event. Thanks for the help, Clint

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add-users scripts
by Adam Cooley 28 May '13

28 May '13

hey gang, new to freeswitch here..I installed freeswitch 1.2.9 via centos, with yum, and was wondering where I would procure the add-user scripts for adding users on the cli? i can't seem to find them on the filesystem.. -adam

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dbh:query - insert id
by Lloyd Aloysius 28 May '13

28 May '13

Hello All How to get the id value after insert a record a record using dbh:query *table_a - columns*. id - auto increment field1 field2 dbh:query("insert into table_a ( field1,field2) values ('11','Test')") After insert how to get the table_a - id value for the inserted record? Thanks Lloyd

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Splitting CDRs on call forward
by Jon Schøpzinsky 28 May '13

28 May '13

Hi List, I am implementing call forwarding on a multi tenant system, and therefore need to split CDR's when the call forward happens, so that if the receiving user also has his account call forwarded, he pays for his part of the call. A calls B B forwards to C C forwards to an external mobile phone. B has a free call from B to C, but C needs to pay for the forwarding to the mobile phone. Therefore i need a separate CDR for the C to Mobile phone call. Another example would be this A works in Company A, and B works in Company B They are both users on our system, and therefore is on the same freeswitch. A calls B B Forwards to an external mobile phone. Here A needs to pay for the call from A to B, and B needs to pay for the call being forwarded to his mobile phone. Do anybody have an idea as to how to implement this in freeswitch. Back in my Asterisk days, this would be done by the ForkCDR command. Venlig hilsen/kind regards Jon Leren Schøpzinsky

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Centralized SIP directory
by mehroz 28 May '13

28 May '13

Hi all, I have a DB server, storing SIP users and auth information. 2 freeswitch servers configured with mod_xml_curl , and an attempt of registration on both servers looks for SIP authentication from DB server. At the moment, i am defining 2 domains on DB server (FS-1 IP and FS-2 IP), and creating unique user , 2 times to address both domains. I want to get rid of creating 2 users for both domains. What is the best approach. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Centralized-SIP-directory-tp7… Sent from the freeswitch-users mailing list archive at Nabble.com.

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FS bypass ZRTP with TURN relay
by adahary 28 May '13

28 May '13

I've setup the FS with bypass_media mode to let endpoints to pass media through a TURN server. The dialplan is set dynamically to bypass_mode=true. The endpoints can now register SIP with FS and then pass media between thenself via the TURN relay (FS not in between) with no one-way-audio issue. The problem is when I setup FS and the endpoints with ZRTP. Even though all RTPs are passing directly between endpoints and FS log indications show that ZRTP is activated on both endpoints (including a=zrtp-hash: in both SDPs) and still the ZRTP is not poping up in the endpoints display. I assume that once media is being relayed between endpoints then ZRTP should pass between like it happens when both endpoints on the same LAN network. What could cause the ZRTP not to work between endpoints when media is bypassing the FS? regards Assaf -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-bypass-ZRTP-with-TURN-rela… Sent from the freeswitch-users mailing list archive at Nabble.com.

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SSL/TLS customized encryption.
by mehroz 28 May '13

28 May '13

Hi all, I am trying to create certificates and keys on a customized encryption scheme. AFAIK gentls_cert generates certificates and keys with : Public Key Info: Public Key Algorithm: rsaEncryption Public-Key: (2048 bit) I want the encryption to be according the SUITE-B standard ,i .e 1. (AES) with key sizes of 128 and 256 bits 2 . Elliptic Curve Digital Signature Algorithm 3 . Elliptic Curve Diffie–Hellman (ECDH) — key agreement 4 .Secure Hash Algorithm 2 (SHA-256 and SHA-384) — message digest I can see FS script (gentls_cert) is using openssl at the back. And as i have very little idea of encryption and all using openssl , i need to know if this is possible to mend this script to our requirements, if so ... how and where should i look at first! Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SSL-TLS-customized-encryption… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Getting CHANNEL_PROGRESS event only sometimes when originating call
by mbo 27 May '13

27 May '13

We are using Freeswitch version 1.2.9. connected with an Sangoma A104 card to the PSTN. If I originate a call though esl socket connection, I'm not always getting a CHANNEL_PROGRESS event. In the test scenario, I'm repeating the originate a couple of times with the same phone number and same parameters, sometimes I get CHANNEL_PROGRESS sometimes not. In the Q.931 trace I can see that I always get a ALERT message, which if I see it correctly, should be converted into a CHANNEL_PROGRESS event. We also have a demo system here with 2 Sangoma A101 cards connected with an crossed E1 cable (also Freeswitch 1.2.9). If I perform the same test here, I always get the CHANNEL_PROGRESS event. Are there any known issues that CHANNEL_PROGRESS event isn't fired reliable when originating calls via freetdm? Thanks Markus

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