add variable in dialplan
Hi list, I want add a sip header in specific variable Ex: ${sip_h_X-Provider-ID} in other variable "Provider" because, I want unset "X-Provider-ID" header in leg-B (command bridge), If I unset "${sip_h_X-Provider-ID}" before bridge command, this command is logicaly not ok => "<action application="bridge" data="sofia/gateway/GW* ${sip_h_X-Provider-ID}*/${destination_number}"/>" I would like : 1) set new variable: ${sip_h_X-Provider-ID} ==> Provider variable 2) unset sip_h_X-Provider-ID 3) bridge with sofia/gateway/GW*${Provider}*/${destination_number} thanks in advance -- Cordialement Hubert Mickaël Hexanet --
Hi all, i use the actual version from git. Sipgate germany is my Provider. After these update, no audio coming out. I have setup the Ivr. The playback application is running, but no audio on the current channel. I send a audiofile with: uuid_broadcast. The same problem. Before i update from: 1.2.5.2 to 1.2.7, Sipgate and all audio ivrs works correctly. I downgrade to: 1.2.5.2 the system works fine. Can your help please? thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste für blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc
Sounds like a possible bug, please report to Jira. Make sure to attach debug logs, pcaps etc. /Peter ________________________________________ Från: freeswitch-users-bounces@lists.freeswitch.org [freeswitch-users-bounces@lists.freeswitch.org] för Thomas Hoellriegel [admin@blindi.net] Skickat: den 7 april 2013 13:34 Till: FreeSWITCH Users Help Ämne: [Freeswitch-users] Problem No audio from ivr using actual git Hi all, i use the actual version from git. Sipgate germany is my Provider. After these update, no audio coming out. I have setup the Ivr. The playback application is running, but no audio on the current channel. I send a audiofile with: uuid_broadcast. The same problem. Before i update from: 1.2.5.2 to 1.2.7, Sipgate and all audio ivrs works correctly. I downgrade to: 1.2.5.2 the system works fine. Can your help please? thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste für blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc !DSPAM:5161561632761574121359!
Hello guys, when i call a number set in my FS using soft client, it works well. However, when i call it with a phone, i get the following: [WARNING] sofia.c:8159 IGNORE INFO DTMF(2) (This channel was not configured to use INFO DTMF!) Any idea? thanks
Hello, Please see the following URLs; http://lists.freeswitch.org/pipermail/freeswitch-users/2011-August/075689.ht... http://freeswitch-users.2379917.n2.nabble.com/DTMF-Info-td6791933.html http://lists.freeswitch.org/pipermail/freeswitch-users/2011-September/076333... This was found using a few simple searches in Google. Please remember that these mailing lists are contributed to with peoples spare time, and you should always try and do your homework / google-fu before asking questions. As a rule of the thumb, try googling for your problem first, if you can't find a solution then post on the mailing list, and be sure to include this in your post (i.e. explain what you have already tried, documentation you have read etc). Cal On Sun, Apr 7, 2013 at 2:54 PM, Sherif Omran <sherifomran2000@yahoo.com>wrote:
Hello guys,
when i call a number set in my FS using soft client, it works well. However, when i call it with a phone, i get the following:
[WARNING] sofia.c:8159 IGNORE INFO DTMF(2) (This channel was not configured to use INFO DTMF!)
Any idea?
thanks
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participants (5)
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Cal Leeming [Simplicity Media Ltd]
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Mickael Hubert
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Peter Olsson
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Sherif Omran
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Thomas Hoellriegel