- FreeSWITCH-users - FreeSWITCH Mailing Lists

Change Gateway Context to receive calls
by Caio Assis 03 Dec '18

03 Dec '18

Hello! I have to configure a Gateway to receive calls from it and process it correctly. The problem is that I haven't found a way to change the context, and, when I receive a call from a Peer with the same IP as the Gateway, FreeSwitch ignores the authentication from the Peer and checks if the IP is allowed in ACL domains. Call comes from authenticated Peer 1000 though context CUSTOM -> OK I have to configure a Gateway which has the same IP as the Peer 1000, so I allow it in ACL. Then, all calls from Peer 1000 come through PUBLIC context. Thanks.

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Diversion Header in Dialplan
by Igor Potjevlesch 03 Dec '18

03 Dec '18

Hello freeswitch users, I’m discovering freeswitch, and I want to understand how to extract and use the variable « sip_h_Diversion » in the Dialplan. I know this question has been asked more times in the mailing-list but I didn’t understand the issue of the diverse topic who evocate this variable. Also I read the information on the wiki but I wasn’t able to know what I have to do in my case. In my Dialplan , I tried to put : <condition break="on-true" expression="^(?:<sip:)?0(\d{9})((?:@.+)?)$" field="${sip_h_Diversion}"> <action application="set" data="sip_h_Diversion=<sip:$1$2"/> </condition> to apply a modification on the Diversion Header, but this condition is unrecognized. I hope someone could explain me, if I have to extract this variable before to use it and how to simply do it ? Thanks, Bests Regards, Igor

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FS not un-registering at shutdown
by David Villasmil 02 Dec '18

02 Dec '18

Hello guys, I was under the impression FS would un-register from gateways when shutting down, is this not the case? I don't see it sending a REGISTER with expires 0. Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

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Register to freeswitch with 2 IPs on the same interface
by Markus Bönke 30 Nov '18

30 Nov '18

Hello all, we have a server with 2 external IPs (eth1 .204 and eth1.1 .210 at the end). The sip profile is listening on .210 and clients are sending REGISTER requests to .210. But in table sip_registrations field hostname and orig_server_host there is the IP .204. Is there a setting to force freeswitch to store the IP where the REGISTER was sent to? Or is there an issue when the IP is configured as second IP on the same interface? Thanks and regards Markus

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Diversion Header in Dialplan
by Alexis DEVIN 30 Nov '18

30 Nov '18

Hello freeswitch users, I’m discovering freeswitch, and I want to understand how to extract and use the variable « sip_h_Diversion » in the Dialplan. I know this question has been asked more times in the mailing-list but I didn’t understand the issue of the diverse topic who evocate this variable. Also I read the information on the wiki but I wasn’t able to know what I have to do in my case. In my Dialplan , I tried to put : <condition break="on-true" expression="^(?:<sip:)?0(\d{9})((?:@.+)?)$" field="${sip_h_Diversion}"> <action application="set" data="sip_h_Diversion=<sip:$1$2"/> </condition> to apply a modification on the Diversion Header, but this condition is unrecognized. I hope someone could explain me, if I have to extract this variable before to use it and how to simply do it ? Thanks, Bests Regards, Alexis Devin

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dialplan xml pre-process include other xml
by Allen Underdown 29 Nov '18

29 Nov '18

I'm having a problem understanding the steps and/or proper formatting of including a xml file inside a "host" xml file within the dial plan. I have limited access to the dial plan, so need to insert a call to a file in a different directory for a customized extension. Example - in /dialplan/somedialplan.xml <?xml version="1.0"?> <document type="freeswitch/xml"> <section name="dialplan" description="Dialplan"> <context name="default"> <extension name="nowhere"> <blah> </extension> </context> </section> </document> I want to add: <X-PRE-PROCESS cmd="include" data="/custom-dir/custom_dial.xml"/> so that every time somedialplan.xml is read, custom_dial.xml is included. I've tried adding it after the section name, and after the context, but it's never parsed (according to watching fs_cli). So I can't seem to figure out where to put it and have it parsed. I'm also a bit confused on exactly the XML required in custom_dial.xml. Do I need to specify the context that I want the extension executed in? In custom_dial.xml I've tried using with and without the <include> tags. I've scoured freeswitch.org but can't seem to find the documentation in a way that I can figure this out. Thanks! Allen Underdown 'Progress isn't made by early risers. It's made by lazy men trying to find easier ways to do something.' --- Robert Heinlein

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RFC2543 from March 1999 called; They want their 0.0.0.0 hold method back.....
by George 29 Nov '18

29 Nov '18

any way to hack around this issue the carrier is not fixing the issue, any calls to an IVR do do throug after the innitial recorded message, this is especially a problem with 311 calls, this is up in Canada with SHAW Thank you, George

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Inability of External SIP Phone to Send IM (Instant Messaging)
by yosin sun 29 Nov '18

29 Nov '18

Dear Sir/Mme, I have a question concerning the inability of an external SIP phone to send IM (Instant Messaging) as follows: An external SIP phone (Zoiper) was able to connect to and talk with another SIP phone, but could not send IM (Instant Messaging). My FreeSWITCH version was 1.6.12. The private IP of my FreeSWITCH was 192.168.1.84 and the public IP was 220.128.137.70. When the external SIP phone (phone number 1001) sent text messages to another SIP phone, the following error message was displayed: sofia_presence.c:225 Can't find registered user 1001(a)220.128.137.70 The current solution was to modify sofia_presence.c:216 from: } else if (!(list = sofia_reg_find_reg_url_multi(profile, user, *host ))) { to: } else if (!(list = sofia_reg_find_reg_url_multi(profile, user, NULL))) {/*ignore host value*/ or another solution was for the SIP phone to register with domain, but it was too troublesome. Is there any solution other than modifying the source code as shown above? Your great help is highly appreciated. Best regards, Jack Su 回覆轉寄 <https://drive.google.com/u/0/settings/storage?hl=zh-TW> <https://www.google.com/intl/zh-TW/policies/terms/> <https://www.google.com/intl/zh-TW/policies/privacy/> <https://www.google.com/gmail/about/policy/>

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Transfer/Merge Call
by Vishal Dalsania 29 Nov '18

29 Nov '18

I have forwarded a call to another gateway and after certain period of time i want to transfer that call to an extension and hangup call to the gateway how can i do that? Basically an outside call lands to FreeSwitch, i forward it to another FreeSwitch at different location but after say 2 minutes i want the call to routed back to internal extension and hangup ongoing call to second FreeSwitch. What is the best way to achieve this?

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odbc issues with Deb9. how FS community members solve them ?
by Yuriy Nasida 29 Nov '18

29 Nov '18

Hi guys, I was very glad to know that Deb9 is supported officially by FS team and did some tests. Unfortunatelly I noted a lot of odbc problems with Deb9. Freeswitch is NOT the reason but I would like to ask other comunity members how they solve such problems. A lot of FS modules can not work with pgsql directly (like mod_lcr for example) and that is why I can not just stop to use odbc. Which issues ? 1) Well, 1st of them is odbc-postgresql in case I try to reach pgsql10. Please look below. I can send odbc.ini file but don't want to send too many info:) # isql -v pgsqldsn DEBUG: CommitTransaction(1) name: unnamed; blockState: STARTED; state: INPROGR, xid/subid/cid: 0/1/0 *Segmentation fault* 2) Second issue with mysql-percona. I suppose everybody knows that libmyodbc.so doesn't exist in Deb9. There is possibility to install manually odbc connector for mariadb but... what should I do for percona ? Also I saw very sad feedbacks from asterisk guys who tried to use mariadb odbc connecter. Please advce that you use to get odbc working with Deb9 Thanks.

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