October 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

SIP Registration Failing
by David Wafula 14 Oct '10

14 Oct '10

Somehow, am unable to register a softphone, no matter what i do. I decided to run wireshark from the machine running the softphone, and so i get the following: 14485 3007.314022 146.141.76.164 146.141.76.153 ICMP Destination unreachable (Port unreachable) 146.141.76.164 is the softphone machine, 146.141.76.153 is the server. This certainly tells me problem, the port, right? But how could it be? when i run sofia status on the server, it shows me , among many things: SIP-IP: 146.141.76.153 URL: sip:mod_sofia@146.141.76.153:5060 BIND-URL: sip:mod_sofia@146.141.76.153:5060 I assume this port 5060 is the problematic one? And, am running these in a LAN, all ports are as open as ever. Thanks again for your help. -- David Wafula

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CCM Interop
by Brian West 14 Oct '10

14 Oct '10

We are looking for a SIP trunk from Cisco Call Manager pointed at one of our Dev boxes so we can do interop. to ensure it works with FreeSWITCH. If you have one can you please find me on IRC bkw_ or email me. Thanks you. /b

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devel-bootstrap.sh - what is it for?
by Jeffrey Leung 14 Oct '10

14 Oct '10

Today when I was pulling the latest revision of freeswitch via git and I noticed git created a new file called devel-bootstrap.sh. I'm wondering what this new devel-bootstrap script does and how different is it from the bootstrap.sh script we're used to when we build freeswitch from sources.

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H323 to Sip conversion problem
by John Carpenter 14 Oct '10

14 Oct '10

Hi, I am trying to bridge an incoming h323 call to an external SIP provider, I am using latest git release. If I use G729 codec I get message "Unsupported ptime of 2 on write Audio codec G.729{sw} for connection [0xb4c0ee10]" and call fails to connect. If I use ulaw codec call fails to connect with "Write PDU fail: no control channel" If I make a straight SIP to SIP call though same provider all work ok. I have posted log of call in http://pastebin.freeswitch.org/14216 because it is rather large. This is my first venture into using the mod_h323 module and maybe I am doing something stupid but have read all docs and seem to have hit a brick wall on this. my h323.conf.xml file looks like this <configuration name="h323.conf" description="H323 Endpoints"> <settings> <param name="trace-level" value="10"/> <param name="context" value="public"/> <param name="dialplan" value="XML"/> <param name="codec-prefs" value="PCMA,PCMU,GSM,G729"/> <param name="use-rtp-timer" value="true"/>  <param name="rtp-timer-name" value="soft"/>    <param name="gk-address" value=""/>  <param name="gk-identifer" value=""/>  <param name="gk-interface" value=""/>  <param name="gk-retry" value="30"/>  <param name="faststart" value="true"/>  <param name="h245tunneling" value="true"/>  <param name="h245insetup" value="true"/>  <param name="jitter-size" value="60"/>  <param name="progress-indication" value="8"/>  <param name="alerting-indication" value="8"/>  <param name="endpoint-name" value="fs"/> <param name="fax-old-asn" value="true"/> </settings> <listeners> <listener name="default"> <param name="h323-ip" value="$${local_ip_v4}"/> <param name="h323-port" value="1720"/> </listener> </listeners> </configuration> And this is the log of the xml_curl dialplan that is executed <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="dialplan" description="php dialplan"> <context name="public"> <extension name="normal"> <condition field="destination_number" expression="^(\d+)$"> <action application="set" data="bypass_media=false"/> <action application="set" data="proxy_media=true"/> <action application="bridge" data="sofia/gateway/arbinet-o/$1"/> </condition> </extension> </context> </section> </document> Any help will be greatly appreciated regards, John Carpenter

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mod_managed socket binding problem
by babak yakhchali 14 Oct '10

14 Oct '10

Hi I'm creating a module in mod_managed to accept snmp request and respond by returning the result of calling some apis. I'm creating a udp socket to listen for snmp reqs. problem is whenever the module is reloaded, because first a new load occurs, in the new one the socket can not be bind, because of the previous socket still running. how can I stop the old socket? thanx

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ACL and Digest authentication problem
by katarina djakovic 14 Oct '10

14 Oct '10

Dear FreeSwitch users, we need some help about ACL and Digest authenication. This is what we want: 1) We want certain users to be authenticated through ACL (certain IP addresses) including both Register and Invite messages. In other words, we want those users to be granted access to our FS withouth having to authenticate with username and password when registering or calling. 2) On the other hand, if users don't fall into our ACL list (registering/calling from other IP addresses) we want them to authenticate normally throught Digest authentication (username/password). We tried to configure FS for our needs, but we didn't acomplished what we wanted. Namely, now, for any users that do not belong to the ACL list our FS will reject their registration and will NOT fall back to Digest authentication. In other words, our FS will let all users that fall into ACL list register and call without authenticating --- but all others will be rejected on the attempt to register (debug trace says: sofia_reg.c IP YY.YY.YY.YY Rejected by register acl "domains") and will not let them fall back to Digest authentication. These are our settings: a) acl.conf.xml: <configuration name="acl.conf" description="Network Lists"> <network-lists>  <list name="domains" default="deny"> <node type="allow" domain="$${domain}"/> <node type="allow" domain="XX.XX.XX.XX/32"/> </list> </network-lists> </configuration> b) sip profile: <param name="apply-inbound-acl" value="domains"/> <param name="apply-register-acl" value="domains"/> <param name="auth-calls" value="true"/> c) users that fall into ACL will have a cidr parameter set aproprietelly <user id="2000" mailbox="2000" cidr="XX.XX.XX.XX/32"> Other users, that we want to be authenticated through Digest authentication will not have anything related to ACL in their user profiles in the Directory. 2) On the other hand, if we remove the <param name="apply-register-acl" value="domains"/> from the sip profile, then users that do not belong to the ACL list will register normally and when calling - their calls (Invite) will fall back to digest authentication (here is the debug: "sofia.c:5847 IP YY.YY.YY.YY Rejected by acl "domains". Falling back to Digest auth.). That is fine with us - but then we have a different problem, then the users from the ACL list will be asked to register by username/password credentials, i.e. their registration will have to authenticated and that is not what we wanted. We are mistaging somewhere. Hopefully what I wrote makes sense and maybe someone could help us configure the system to fit our needs. Many thanks in advance, Katarina

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Fwd: Fwd: [Rates] Please tell Meftah Tayeb to stop harass and damage my company
by Meftah Tayeb 14 Oct '10

14 Oct '10

ban diegoviola piss of shit of the world -------- Message original -------- Sujet: Fwd: [Rates] Please tell Meftah Tayeb to stop harass and damage my company Date : Wed, 13 Oct 2010 18:13:09 -0400 De : Yaro Donchenko <yaro(a)ovetel.com> Pour : Tayeb Meftah <tayeb.meftah(a)gmail.com> ---------- Forwarded message ---------- From: *Diego Viola* <diego.viola(a)gmail.com <mailto:diego.viola@gmail.com>> Date: Wed, Oct 13, 2010 at 5:46 PM Subject: [Rates] Please tell Meftah Tayeb to stop harass and damage my company To: rates <rates(a)ovetel.com <mailto:rates@ovetel.com>>, we-sales(a)ovetel.com <mailto:we-sales@ovetel.com>, ee-sales(a)ovetel.com <mailto:ee-sales@ovetel.com>, cis-sales(a)ovetel.com <mailto:cis-sales@ovetel.com>, mena-sales(a)ovetel.com <mailto:mena-sales@ovetel.com>, africa-sales(a)ovetel.com <mailto:africa-sales@ovetel.com>, Tony Volkov <tony(a)ovetel.com <mailto:tony@ovetel.com>> Hello OVETEL, This person that works for your company, Tayeb Meftah, it's an homosexual freak that harass me every day in the Messenger, and I believe he has serious mental issues. I believe he is trashing the image of your company. Please ask him to stop bothering me and tell stupid things about me and my company. This freak is trashing your image and of your company, so if you are interested to have him in your company, it's your problem. But I just want him to stop bothering me and stop saying bullshit about me and my company. For you to see that this is all true, I put at your disposal the conversations that I had with the freak of your company. When he told me that he was the owner of OVETEL and when he told me that he told you bad things about me and my company’s image. This homosexual freak of your company is obsessed to write me and ask me things for free. Please take matters into this subject, and if you wish to have this person in your company it’s your problem. If this freak continues to harass me and damage the name of my company I will have to take judicial actions against him so please take some control about this. Best Regards, Diego Viola Representative of Bridgecom LLC ---------- Forwarded message ---------- From: tayeb(a)ovetel.com <mailto:tayeb@ovetel.com> <tayeb(a)ovetel.com <mailto:tayeb@ovetel.com>> Date: Wed, Oct 13, 2010 at 6:24 PM Subject: Chat with tayeb(a)ovetel.com <mailto:tayeb@ovetel.com> To: diego.viola(a)gmail.com <mailto:diego.viola@gmail.com> 5:47 PM tayeb.meftah: ovetel don't speak with you anymore just don't wait for it 6:18 PM me: what did you tell them? 6:19 PM tayeb.meftah: shut up and go away from me me: ok tayeb.meftah: ecause you are montaly infected and you have a very high late 6:20 PM and you lie in adition to this me: ? you are an idiot for doing this tayeb.meftah: and you? go fuck yourself 11:48 AM tayeb.meftah: http://www.ovetel.com my company 11:49 AM me: nice are you the owner of this company? tayeb.meftah: me and other one Notice: This mail is covered by the Electronics Communications Privacy Act, 18 U.S.C. 2510-2521 and is legally privileged. This message, together with any attachments, is intended only for the addressee. It may contain information which is legally privileged, confidential and exempt from disclosure. If you are not the intended recipient, you are hereby notified that any disclosure, copying, distribution, use or any action or reliance on this communication is strictly prohibited. If you have received this e-mail in error, please notify the sender immediately by telephone +1-212-401-0707 or by return e-mail to the original sender and delete and permanently erase this message and all attachments of any and all originals and copies. -- Best Regards *Yaro Donchenko* /VP of Business Development/ Direct: +1-845-475-9347 Mobile: +1-917-267-9276 MSN: yaro(a)ovetel.com <mailto:yaro@ovetel.com> Yahoo: yaroovetel OVETEL USA: +1-212-401-0707 OVETEL UK: +44-20-34110486 OVETEL Ukraine: +380-44-3607309 OVETEL Jordan: +962-6-2508905 OVETEL South Africa: +27-11-4613345 Organize a meeting with me: http://tungle.me/yaro Follow us on Twitter for the latest price updates: http://twitter.com/ovetel Notice: This mail is covered by the Electronics Communications Privacy Act, 18 U.S.C. 2510-2521 and is legally privileged. This message, together with any attachments, is intended only for the addressee. It may contain information which is legally privileged, confidential and exempt from disclosure. If you are not the intended recipient, you are hereby notified that any disclosure, copying, distribution, use or any action or reliance on this communication is strictly prohibited. If you have received this e-mail in error, please notify the sender immediately by telephone +1-212-401-0707 or by return e-mail to the original sender and delete and permanently erase this message and all attachments of any and all originals and copies.

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3-way calling fails with git
by Woody Dickson 14 Oct '10

14 Oct '10

Hi, I am having problem with using the att_xfer app after upgrading to the latest git. After the callee presses a meta_app key and bridged a 3rd party in, the callee got hung up immediately leaving the 3rd part and the original caller talking to each other only. I was able to get 3-way call to work in the previous release of fs. Does anyone know if this is a bug or a bad config on my end? Thanks, Woody

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fifo help
by Kyle Haefner 14 Oct '10

14 Oct '10

Hi All, I'm pretty new to freeswitch :) I'm trying to set up the following scenario: 1. A caller calls in is greeted by a message that says, "Please hold while your party is located" 2. The caller is placed on hold 3. An outbound call sequentially rings cell phones through a gateway 4. If a cellphone answers the "agent" must push a digit (to verify they are not voice mail) 5. If a digit is pushed the agent is connected to the caller I'm trying to implement this using mod_fifo, however when I add the caller to the fifo, processing of that extension stops and I can't ever call and test the cellphone to add them as a member of the fifo. <context name="custom"> <extension name="FIFO-Call"> <condition field="destination_number" expression="^(100)$"> - <action application="answer"/> <action application="playback" data="/usr/local/freeswitch/sounds/en/us/callie/misc/noc_intro.wav"/> <action application="fifo" data="noc_fifo in nowait"/> <action application="sleep" data="300"/> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/misc/prove_youre_human.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="set" data="fifo_member(add noc_fifo {fifo_member_wait=nowait}sofia/custom_dialplan/89883757@pstn-gateway :5060)"/> </condition> </extension> </context> Any help would be greatly appreciated! Kyle

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Multiple Soundcard Channels
by Erik M. Devane 13 Oct '10

13 Oct '10

[Repost - apologies to all for my earlier unintentional hijack.] Can I start multiple instances of mod-portaudio? I would like to have multiple channels from my alsa-compliant soundcard but I can't find a way to run multiple channels in FreeSWITCH. I am connecting to a PABX and need to bridge calls. Am I missing something obvious? Thank you, Erik

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