[asterisk-users] sip can not transmit fax receive from chan dahdi
hello every body
i have problem in receiving fax from e1 lines. this is my scenario:
faxphoneericson pbx ---e1asterisksip-zoiper-softphone
when i send fax from zoiper, i can receive it successfully on the faxphone
but when i send fax from faxphone, i can not receive it on zoiper. i want
do this just by using faxdetect option. when fax comes from faxphone, chan
dahdi on asterisk detect fax and redirect to fax extension. in fax
extension, i just dial sip peer which is connected to zoiper like this:
exten=fax,1,Dial(SIP/peer-1/${EXTEN})
peer-1 is a sip peer which is defined in sip.conf like this:
[peer-1]
host=192.168.0.XX
type=peer
context=from-trunk
insecure=port,invite
i set debug messages and understand that sip channel can not identify fax
and try to send it like a voice call (set rtp and other things exact a
voice call). what is the problem? does chan dahdi should send something for
sip to identify fax session? or something is wrong related to chan sip?
i struggle a lot but can not solve this problem. any comments or hints are
really appreciated.
SAM
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[asterisk-users] does chan dahdi supports fax?
hello everyone, i have question about fax detection on dahdi channels. does dahdi channels detect fax and pass it? if yes, does it detects both types of fax (g711 pass through and T.38)? finally, how can i enable it on dahdi_channels? i set faxdetect=both in chan_dahdi.conf but dahdi can not pass fax(i just wanna pass fax not send or receive it). any comments or hints are really appreciated. SAM -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] reduce delay in fax detection
hello everybody i want to send fax via asterisk in pass through mode. everything is ok if enable fax detection in ooh323 and write fax extension in extensions.conf file. just one problem: delay. i have to wait 5 seconds in order to fax detection done. it is too long for me when i have voice call and no fax. my phone rings after five seconds. is there any way to omit or reduce this time? i test and understand that sip fax detection acts in some milliseconds but oohs323 needs 5 seconds to do that. what is difference between them? any comments or hints are really appreciated. SAM -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] reduce delay in fax detection
hello and thank you so much for your reply just one question: how do you use it? AFAIK, when asterisk receive a call, it select extensions based on the received number and when it detects fax, jump to the fax extension. now when should i use these show commands to detect fax and how should i tell asterisk to execute fax extension? this is a big problem for me, so i really appreciate if you help me to solve it. yours, SAM On Wed, May 20, 2015 at 7:11 PM, Tech Support aster...@voipbusiness.us wrote: Hey; Yes, I’ve also seen that 5 second delay with our fax server and it drove me crazy. How I solved it was by doing a “core show channels concise|verbose” and detect if there was a fax transmission going on. Doing it this way shows up instantaneously without any delay. Like so: mte6*CLI core show channels concise SIP/SIPRoutes-0054!faxserver-tx!fax!11!Up!SendFAX!/var/spool/asterisk/fax/documents/faxadmin/default.tif,dfz!!voipbusiness!!3!4!(None)!1432132315.97 mte6*CLI core show channels verbose Channel Context ExtensionPrio State Application Data CallerIDDuration Accountcode PeerAccount BridgedTo SIP/SIPRoutes-00 faxserver-tx fax11 Up SendFAX /var/spool/asterisk/fax/d 00:00:10 voipbusines (None) 1 active channel 1 active call 62 calls processed mte6*CLI core show channels Channel Location State Application(Data) SIP/SIPRoutes-00 fax@faxserver-tx:11 Up SendFAX(/var/spool/asterisk/fa 1 active channel 1 active call 62 calls processed All of these outputs shows that there is a fax transmission taking place. I hope this helps. Regards; JV *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *s m *Sent:* Wednesday, May 20, 2015 2:54 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] reduce delay in fax detection hello everybody i want to send fax via asterisk in pass through mode. everything is ok if enable fax detection in ooh323 and write fax extension in extensions.conf file. just one problem: delay. i have to wait 5 seconds in order to fax detection done. it is too long for me when i have voice call and no fax. my phone rings after five seconds. is there any way to omit or reduce this time? i test and understand that sip fax detection acts in some milliseconds but oohs323 needs 5 seconds to do that. what is difference between them? any comments or hints are really appreciated. SAM -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] can ooh323 work with cisco router?
hello every body, i have big problem to configure h323 trunk between cisco router and asterisk 11.13.1 which uses ooh323 module. any body knows if ooh323 module can work with cisco routers or not (in gateway mode, it is ok and register in cisco gatekeeper but i can not configure trunk h323) any comments or hints are really appreciated. SAM -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can ooh323 work with cisco router?
, ooh323c_o_3) 10:42:10:894 Looking for matching capabilities. (outgoing, ooh323c_o_3) 10:42:10:894 Created new logical channel entry (outgoing, ooh323c_o_3) 10:42:10:895 Built OpenLogicalChannel-OO_G711ALAW64K (outgoing, ooh323c_o_3) 10:42:10:895 Building Facility message for tunneling OOOpenLogicalChannel (outgoing, ooh323c_o_3) 10:42:10:895 Sending Q931 message (outgoing, ooh323c_o_3) 10:42:10:895 Sent Message - Facility(OOOpenLogicalChannel) (outgoing, ooh323c_o_3) 10:42:10:895 Tunneled Message - OpenLogicalChannel(1003). (outgoing, ooh323c_o_3) 10:42:21:947 H.225 Release Complete message received (outgoing, ooh323c_o_3) 10:42:21:947 Cause of Release Complete is 10. (outgoing, ooh323c_o_3) 10:42:21:947 Closing H.245 connection (outgoing, ooh323c_o_3) 10:42:21:947 In ooEndCall call state is - OO_CALL_CLEARED (outgoing, ooh323c_o_3) 10:42:21:947 Cleaning Call (outgoing, ooh323c_o_3)- reason:OO_REASON_REMOTE_CLEARED 10:42:21:947 Clearing all logical channels (outgoing, ooh323c_o_3) 10:42:21:947 Stopped Receive channel 1001 (outgoing, ooh323c_o_3) 10:42:21:947 ERROR: No Open LogicalChannels - Failed FindLogicalChannelByChannelNo (outgoing, ooh323c_o_3 10:42:21:947 Removing call 80494a048: ooh323c_o_3 10:42:21:947 Removed call (outgoing, ooh323c_o_3) from list 10:42:21:947 Ending Call Monitor thread cisco debug shows rtp message with src address 0.0.0.0. i really don't know how i should fix it. please help me. thanks SAM On Wed, May 6, 2015 at 10:44 AM, Dmitry Melekhov d...@belkam.com wrote: 06.05.2015 10:06, s m пишет: hello every body, i have big problem to configure h323 trunk between cisco router and asterisk 11.13.1 which uses ooh323 module. any body knows if ooh323 module can work with cisco routers or not (in gateway mode, it is ok and register in cisco gatekeeper but i can not configure trunk h323) we use chan_ooh323 with cisco for long time with some issues, but... only issue is not solved in current asterisk version is https://issues.asterisk.org/jira/browse/ASTERISK-24400 so you have to be more specific in your big problem description ;-) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can ooh323 work with cisco router?
hello thanks Dmitry for your useful hints. i enable debug and solve my problem:). it was codec compatibility problem. but it is so strange; if i set codec g711alaw in cisco router and asterisk, i have the mentioned problem but if i set codec to transparent in cisco router, every thing will be ok. is there any difference between g711 codecs which cisco and asterisk utilize? On Wed, May 6, 2015 at 11:56 AM, Dmitry Melekhov d...@belkam.com wrote: 06.05.2015 10:58, s m пишет: Hello! I'm not h323 expert, may be somebody else can understand from this log what is happening, but I can't :-( Could you, please, provide log with tracelevel=6 in ooh323.conf ? Thank you! hello Dmitry thank you for your reply. Ok, you are right. i want to configure trunk h323 between asterisk 11.13.1 and 2800 cisco router. this is my scenario: PBX(100)---cisco---asterisk11.13.1PBX(200) when i call from 100 to 200, everything is ok but when i call from 200 to 100, phone rings but after i answer it, i have no voice and call terminates after 5 seconds. this is ooh323 debug(in asterisk11.13.1 system): ooh323_get_rtp_peer OOH323/peer-2-5 - (null):0, 1 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem in h323 trunk to cisco router
hello every body, i want to have h323 trunk between cisco 2800 and asterisk 11.13.1 with ooh323 module. i configured both side and have successful call from cisco to asterisk. but when call comes from asterisk to cisco, my phone rings but no audio is heard and call is disconnected after 5 second. i enable debug voice rtp in cisco and see the source address for receiving rtp packets is 0.0.0.0 Apr 28 07:46:34.765: RTP(50493): ps rx s=0.0.0.0(0), d=192.168.0.139(17112), pt=8, ts=BF40, ssrc=2C1690C9 any body knows how should i fix it? this is my ooh323.conf file: [general] port=1720 context=from-trunk gatekeeper=DISABLE bindaddr=192.X.X.X disallow=all allow=all AcceptAnonymous=yes directrtpsetup=yes directmedia=yes faststart=yes h245tunneling=yes mediawaitforconnect=yes tos=lowdelay [sam] type=user host=192.X.X.X directmedia=yes [sam-1] type=peer host=192.X.X.X directmedia=yes any comments or hints are really appreciated. SAM -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip trunk to Cisco router
hello everybody, i want to configure a sip trunk between my system which has asterisk 11.5.1 and a cisco router. this is my scenario: Freepbx-my system-cisco-routerFreepbx my system acts like a router. if i set just one codec in dial-peers on cisco router, every thing is ok and i can make a call. but if i set different codecs in a voice class codec and assign it to dial-peers in cisco router, i can not make calls. if i change my scenario like: Freepbx--cisco-router--Freepbx calls are succeed without any problem. Freepbx are asterisk-base too, so i think something is wrong in my system (my asterisk configuration is not correct or something is missing). any body knows how should i fix this problem? any comments or hints are really appreciated. P.S: my sip.conf: [peer-1] host=X.X.X.X type=peer context=from-trunk allow=all qualify=yes insecure=port,invite [peer-2] host=Y.Y.Y.Y type=peer context=from-trunk allow=all qualify=yes insecure=port,invite -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] h235 for authenticating RAS message
hello everybody, i want to have authentication on RAS messages between gatekeeper and gateway. i have a cisco gatekeeper and an asterisk gateway. is it possible to have h235 on asterisk gateway in order to send authenticated RAS message to gatekeeper? if yes, how can i add it to my asterisk? i am using 00h323 module for mu h323 connections. any comments or hints are really appreciated. SAM -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ! in dial-pattern not work with overlap dialing
hello all i want to have overlap dialing in asterisk. it works fine if i don't have ! in my pattern. for example pattern 07. works fine and i can call 07122 by overlap dialing via it. but if i define 07! i can't call 07122 because it doesn't wait to collect all digits and therefore call 07. if i define 07.! it works fine again and wait to collect all digits and hence 07122 rings. what is wrong with my patterns? how should i use ! in my pattern in order to have overlap dialing? every comment or points are appreciated. SAM -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how determine mandatory modules to slimming asterisk
hello guys i want to slimming my asterisk by loading only mandatory modules. in order to do that, i edit my modules.conf file and set autoload=no and load just mandatory modules. my problem is, how should i determine which modules are necessary to asterisk works correctly? i have sip, h323 and dahdi connection on my asterisk. is there any documentation about mandatory modules for asterisk? or anybody has such a list? any comments or hints are appreciated SAM -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] set different codec for different sip calls
hello every one i want to have multiple sip calls with different codecs for each one. for example call to 8100 has g729 codec while call to 7900 has ulaw codec. i searched a lot and found that there is some variable like sip_codec which can set codec for a special inbound or outbound call. i don't try it yet because i prefer to set the codec for each call by setting it in contexts in sip.conf or sip_additional.conf file. is it possible?? if yes, how should i set codec for each context? if not, setting the codec in dial-plans in extensions.conf file, is the only way??? thanks in advance SAM -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how apply new configuration to ooh323 without disconnecting current calls
hello all i'm using ooh323.so module for my h323 connections and it works fine. i just have problem with loading and unloading module. you know, ooh323 module doesn't support reload command. it means, if ooh323 module is loaded and i reconfigure my h323 channels (add another channel), i should unload and load module again. it causes to disconnect all h323 connections which are connected before and it is not good for me at all. i want to know if there is any solution to add some configuration to h323 and apply them without disconnecting current connected channels. thanks in advance SAM -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is g729 codec free? or under license???
thanks Asghar, but are you sure? my two endpoints -which are soft-phones- have g729 codec but my asterisk on middle system has not any module for g729 codec. i think i should get module g729 for my middle system in order to pass calls with g729 codec. isn't it true? On Sat, Oct 12, 2013 at 1:08 PM, Asghar Mohammad asghar...@gmail.comwrote: HI, You don't need a g729 installed in pass throw mode. if both ends have codec g729 you can just enable on both peers. and asterisk should pass the codec from 1 end to other. but make sure you are not doing transcoding of any type answering the call playing voice prompts etc. On Sat, Oct 12, 2013 at 9:52 AM, s m sam.gh1...@gmail.com wrote: thank you everybody for your useful replies and so sorry to answer late. i understand what i need. first of all, i wanna to use pass through g729 codec (which is free). so i go to http://asterisk.hosting.lv/ to get g729 codec. i have freebsd 8.2 and asterisk 1.8.22 but there is no compatible codec for Xeon Intel in the list. it means that i can't use codec g729 on my system??? or can i use codec for another type of hardware for my system? anyone has any experience? thanks in advance SAM On Mon, Oct 7, 2013 at 5:04 PM, John Novack jnov...@stromberg-carlson.org wrote: Darryl Moore wrote: Thank you Steve, and I read a bit more on the web on this subject including your own well reasoned page at http://www.soft-switch.org/**patents/index.htmlhttp://www.soft-switch.org/patents/index.html However, despite wide acceptance of the patentability of such codecs (unfortunately), whether they are in fact software patents or not appears to be a matter of opinion. The FSF and Fedora both refer to codec patents as being software patents. http://endsoftpatents.org/**2011/02/usa-patent-reform-not-**enough/http://endsoftpatents.org/2011/02/usa-patent-reform-not-enough/ http://fedoraproject.org/wiki/**Software_Patentshttp://fedoraproject.org/wiki/Software_Patents A quick google search of both terms will show that there are a great many people who see codec patents as software patents, so I don't think I am alone there. snip Law is ALWAYS open to interpretation, so that is not surprising. See if you can get any lawyer, and especially a patent attorney, to give you a definitive answer! You will not get one. Seldom will you ever get an eggspurt legal opinion Any good lawyer will tell you maybe, or if there is any doubt don't do it! Law is not precisely measurable. No meter or O'scope to assist here. Any A**hole can sue anyone for the filing fee, and the results are up to the opinion of a judge or jury. The lawyers want it that way, so it isn't ever going to be any different. John Novack -- Dog is my Co-pilot -- __**__** _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is g729 codec free? or under license???
thank you everybody for your useful replies and so sorry to answer late. i understand what i need. first of all, i wanna to use pass through g729 codec (which is free). so i go to http://asterisk.hosting.lv/ to get g729 codec. i have freebsd 8.2 and asterisk 1.8.22 but there is no compatible codec for Xeon Intel in the list. it means that i can't use codec g729 on my system??? or can i use codec for another type of hardware for my system? anyone has any experience? thanks in advance SAM On Mon, Oct 7, 2013 at 5:04 PM, John Novack jnov...@stromberg-carlson.orgwrote: Darryl Moore wrote: Thank you Steve, and I read a bit more on the web on this subject including your own well reasoned page at http://www.soft-switch.org/**patents/index.htmlhttp://www.soft-switch.org/patents/index.html However, despite wide acceptance of the patentability of such codecs (unfortunately), whether they are in fact software patents or not appears to be a matter of opinion. The FSF and Fedora both refer to codec patents as being software patents. http://endsoftpatents.org/**2011/02/usa-patent-reform-not-**enough/http://endsoftpatents.org/2011/02/usa-patent-reform-not-enough/ http://fedoraproject.org/wiki/**Software_Patentshttp://fedoraproject.org/wiki/Software_Patents A quick google search of both terms will show that there are a great many people who see codec patents as software patents, so I don't think I am alone there. snip Law is ALWAYS open to interpretation, so that is not surprising. See if you can get any lawyer, and especially a patent attorney, to give you a definitive answer! You will not get one. Seldom will you ever get an eggspurt legal opinion Any good lawyer will tell you maybe, or if there is any doubt don't do it! Law is not precisely measurable. No meter or O'scope to assist here. Any A**hole can sue anyone for the filing fee, and the results are up to the opinion of a judge or jury. The lawyers want it that way, so it isn't ever going to be any different. John Novack -- Dog is my Co-pilot -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is g729 codec free? or under license???
thank you Dominik you help me a lot. and the last question is how many license key should i buy? i read that license for g729 is per-channel but i don't understand what channel exactly means here. this is my scenario : 10endpointspbx181...pbx182...pbx183...10endpoints pbx181 and pbx183 has 10 endpoints connected to them. the call between these endpoints are established by pbx182. if i want to buy a license for pbx182, how many license key do i need? just one because i have just one connection on it? or two, because two trunks is defined on it? or as many as endpoints which are connected to each other via pbx182? please help me to clarify channel concept in my mind. thanks in advance SAM On Tue, Oct 1, 2013 at 11:34 AM, Dominik George n...@naturalnet.de wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA512 Hi, about g729, you mean if it get free g729 and all my systems (PBXs and routers) use g729 codec for setting a call, call is set without any problem? Yes, if all systems use g729 directly, you are ready to go. - -nik -BEGIN PGP SIGNATURE- Version: APG v1.0.8-fdroid iQFMBAEBCgA3BQJSSoHxMBxEb21pbmlrIEdlb3JnZSAobW9iaWxlIGtleSkgPG5p a0BuYXR1cmFsbmV0LmRlPgAKCRAvLbGk0zMOJYmRB/USyTbAqhAsnFZSGGjIcLK7 uQ3nsVNGcmE18LaBN/XFicwp5UjVB5Euju+fjKu1FhqAzECsAPMup/1JUytikmYz +32wV5YL1SNKMA/ddi/zvVa9qIbKA9yP1HuBilpD+W0DO3hdnzr2xrdR1S2z5PGZ pnYWsVlXbWYEslOuK1oaMqINoxWbsQulwQi86GPTCwPtZmhcLrvBm1sDFxWb/oPP lsPy33ZH5BeQ/XEf6nWfoiEu4Hk2S0brCH74zsz9uD6PKL1CFdLcpWv/4k5M+Mly At2PC+leZZ/TX3VNqbasslQkyv/QLZIQVtG0qQ7DGflnkrzNi5/pNV7CVT5sdPQ= =5rja -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is g729 codec free? or under license???
thanks guys for your replies. no, these endpoints are soft phones which can have different codec. now, for my scenario, how many license key in needed? On Wed, Oct 2, 2013 at 11:23 AM, Mitul Limbani mi...@enterux.in wrote: Are these end points Hard IP Phones having g729 codec? If yes then you dont need any license. Just download passthrough g729 license. Mitul On Oct 2, 2013 1:17 PM, Frederic Van Espen frederic...@gmail.com wrote: On 10/02/2013 09:33 AM, s m wrote: and the last question is how many license key should i buy? i read that license for g729 is per-channel but i don't understand what channel exactly means here. this is my scenario : 10endpointspbx181...**pbx182...pbx183...10endpoints pbx181 and pbx183 has 10 endpoints connected to them. the call between these endpoints are established by pbx182. if i want to buy a license for pbx182, how many license key do i need? just one because i have just one connection on it? or two, because two trunks is defined on it? or as many as endpoints which are connected to each other via pbx182? AFAIK, you need one license for each channel that is transcoding from one given codec to g729 (or the other way around). So if at any given time on an asterisk box you would have a maximum of 3 simultaneous calls that are g729 at one end and ulaw at the other, you would need a license key for 3 transcoding channels. Anyone, please correct me if I am wrong on this. Frederic -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] is g729 codec free? or under license???
hello all, i have problem in using g729 codec. my asterisk version is 1.8.22. when i run core show codecs in asterisk, there is a g729 codec in the list so i assume that i can use it for my channels. but connection can not be set when i use it for my h323 channel. i read somewhere that codec g729 is a commercial codec and i should buy its license in order to use it. is it true? if yes, why is it listed in codecs in asterisk?? thanks in advance, SAM -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is g729 codec free? or under license???
thanks Dominik, you're right. i don't pay attention enough about my subject. about g729, you mean if it get free g729 and all my systems (PBXs and routers) use g729 codec for setting a call, call is set without any problem? On Tue, Oct 1, 2013 at 10:35 AM, Dominik George n...@naturalnet.de wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA512 s m sam.gh1...@gmail.com schrieb: hello all, i have problem in using g729 codec. my asterisk version is 1.8.22. when i run core show codecs in asterisk, there is a g729 codec in the list so i assume that i can use it for my channels. but connection can not be set when i use it for my h323 channel. i read somewhere that codec g729 is a commercial codec and i should buy its license in order to use it. is it true? if yes, why is it listed in codecs in asterisk?? thanks in advance, SAM -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi, Asterisk has a free codec for g729, but it can only pass through g729 channels. If you need to translate from/to g729, you need a license from Digium. On a side note, free or under license is nonsense. Free software also has a license; you always need a license to use software - some copyright holders just do not make you pay for it and do not limit your rights (a concept called public domain does not exist globally). - -nik - -- Diese Nachricht wurde von meinem Android-Mobiltelefon mit K-9 Mail gesendet. -BEGIN PGP SIGNATURE- Version: APG v1.0.8-fdroid iQFNBAEBCgA3BQJSSnRBMBxEb21pbmlrIEdlb3JnZSAobW9iaWxlIGtleSkgPG5p a0BuYXR1cmFsbmV0LmRlPgAKCRAvLbGk0zMOJXFJB/9PNVAlAjmPbFN2vGNfzS4r SAIZHfrVWmflxBGFAbVnZriN3Tvw2+haS0/8qV+rRhaK+IdqrF8Jb491a20NPTLu H87tRNTytXwJ94ngWyvTVWbPtqaWuEi8rle40VeYo9qv2xXx5XqFDCeWaTUDol/H 4BXXh4+bnmmKrud9K3gtbFx9hyeKdzcYHjv6WyHjRcQo8lTrOvloR7Pbqk38lIf2 MxFB1E5CKJPmq340xcS80N2K5PS9RFtQlUKDO2r9YaAl3CYz0VRUKC63cLo0Rsdk CBvehGTi85c85qdDXCVKRPUcUWQMDqpNgSRrXifPj6laWb/hciNRweEDf6ATzb8z =k0HG -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] caller id not shown
hello all
i have asterisk 1.8.22 and have problem with caller id. this is my
scenario:
PSTN -- FXO --- FXS --- phone(223)
when i call from a 223 to another phone, every thing is ok and caller id
(223) is shown in called phone. but when i call from another phone to 223,
no caller id is shown and just zero is shown.
if i set callerid=12345 in chan_dahdi.conf file, when another phone call
223, this number (12345) is shown as caller id instead of zero. but i want
to show incoming number as caller id.
this is my chan_dahdi.conf file:
[channels]
;cidsignalling=dtmf
cidstart=polarity;; in gozine takhir dar tamas (aghab boodan yek zang) ra
az beyn mibarad.
callprogress=yes
usecallerid=yes
hidecallerid=no
callwaiting=no
transfer=yes
echocancel=yes
echotraining=yes
callerid=asreceived
group=0
callgroup=1
pickupgroup=1
usecallerid=yes
context=pstn-channels
channel=5-8
group=1
callgroup=1
pickupgroup=1
usecallerid=yes
context=phone-channels
channel=1-4
and this is my extensions.conf file:
[phone-channels]
exten=_.,1,Dial(DAHDI/8/${EXTEN})
[pstn-channels]
exten=_.,1,Dial(DAHDI/2/${EXTEN})
i searched a lot but found nothing useful:( please help me to solve it.
thanks in advance,
SAM
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[asterisk-users] have two H323 connection: one with GK, one with other GW. is it possible?
hello all, i have a conceptual question. i have a h323 gateway and it is connected to a h323 gatekeeper. my question is: can i connect my gateway to another gateway directly? i mean can these two gateways work with each other without working with gatekeeper? or when i have connection with a gatekeeper, all calls must be set through it? thanks in advance, SAM -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
hello all, i want to have ooh323 connection between asterisk and cisco. in my scenario, asterisk is gateway and cisco is gatekeeper. this is my ooh323.conf file: [general] port=1720 bindaddr=192.168.0.227 gateway=yes faststart=yes h245tunneling=yes h323id=g...@test.com settracelevel=10 gatekeeper=192.168.0.212 context=from-trunk disallow=all allow=ulaw allow=alaw allow=gsm dtmfmode=rfc2833 with this config, gateway is registered in cisco gatekeeper correctly. but when i want to call from it, cisco reject my gateway and h225 asn1 messages say incomplete address. i searched a lot and understand that, if a cisco router acts as gateway, it sends h323-id as well as dialed number for gatekeeper but my gateway(which is asterisk), only send dialed number. therefore cisco gatekeeper doesn't know how route this call and reject it. if i define e164 number in ooh323.conf file, every thing is ok and call routed correctly. my question is: can asterisk work with cisco gatekeeper just by h323-id? if yes, how i can do this? in the other words, is it necessary to define e164 number in ooh323.conf file to have a correct connection or not? thanks in advance SAM -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] is necessary to define e164 number in h323 gateway?
hello all, i want to have ooh323 connection between asterisk and cisco. in my scenario, asterisk is gateway and cisco is gatekeeper. this is my ooh323.conf file: [general] port=1720 bindaddr=192.168.0.227 gateway=yes faststart=yes h245tunneling=yes h323id=g...@test.com settracelevel=10 gatekeeper=192.168.0.212 context=from-trunk disallow=all allow=ulaw allow=alaw allow=gsm dtmfmode=rfc2833 with this config, gateway is registered in cisco gatekeeper correctly. but when i want to call from it, cisco reject my gateway and h225 asn1 messages say incomplete address. i searched a lot and understand that, if a cisco router acts as gateway, it sends h323-id as well as dialed number for gatekeeper but my gateway(which is asterisk), only send dialed number. therefore cisco gatekeeper doesn't know how route this call and reject it. if i define e164 number in ooh323.conf file, every thing is ok and call routed correctly. my question is: can asterisk work with cisco gatekeeper just by h323-id? if yes, how i can do this? in the other words, is it necessary to define e164 number in ooh323.conf file to have a correct connection or not? thanks in advance SAM -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] define extension to send calls to gatekeeper
hello every one,
i have an asterisk system and want to act as gateway and send calls to
cisco gatekeeper.
this is my h323.conf file:
[general]
port=1720
binaddr=192.168.0.YY
context=from-trunk
faststart=yes
h245tunneling=yes
gatekeeper=192.168.0.XX //cisco address
progress_setup=8
progress_alert=8
dtmfmode=rfc2833
jbenable=yes
jbforce=no
jbmaxsize=200
jbresyncthreshold=1000
jbimpl=fixed
jblog=no
and this is extension that i defined for it in extensions.conf:
exten=_2.,1,Dial(H323/2${EXTEN:1})
(if i define my extension like
(exten=_2.,1,Dial(H323/to-cisco/2${EXTEN:1}), asterisk returns core dump
error.
but, i can't send my calls to cisco gatekeeper. do you have any suggestion
what is wrong with my configuration?
thanks in advance
SAM
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[asterisk-users] how send calls to gatekeeper?
hello everyone
i have a simple question: i have an asterisk which is a h323 gateway
and has a h323 connection to a cisco gatekeeper and a sip connection
to a pbx.
my question is: how can i send all calls to gatekeeper?
i searched a lot and found that i should set gatekeeper=192.168.0.X
(ip address of my gatekeeper) in h323.conf file.
but what about extensions.conf file? should i define an extension like
a simple h323 connection to gatekeeper (like
exten=_2.,1,Dial(H323/${EXTEN}@cisco_out,60,))? or no dial pattern
need to be defined in extension.conf file? if we should define dial
pattern, what is different between a simple trunk h323 connection and
a gateway-gatekeeper h323 connection?
thanks in advance
SAM
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Re: [asterisk-users] h323-sip: one way connection
oh yes, i'm using h323 not openh323
On Fri, Apr 26, 2013 at 1:16 AM, Asghar Mohammad asghar...@gmail.comwrote:
nuFone h323 or openh323?
On Thu, Apr 25, 2013 at 9:33 PM, s m sam.gh1...@gmail.com wrote:
flavor? i do not understand what you mean. please explain more.
thanks
On Wed, Apr 24, 2013 at 8:16 PM, Asghar Mohammad asghar...@gmail.comwrote:
what flavor of h323 you are using?
On Wed, Apr 24, 2013 at 8:50 AM, s m sam.gh1...@gmail.com wrote:
thanks Asghar,
i do it, but no thing happened:(
asterisk do not identify host line as ip address of the other end
On Tue, Apr 23, 2013 at 9:15 PM, Asghar Mohammad
asghar...@gmail.comwrote:
try type=peer instead of friend.
On Tue, Apr 23, 2013 at 10:04 AM, s m sam.gh1...@gmail.com wrote:
i know what is the exactly problem. i enable debug for h323 and it
says:
could not find user by name 200 or address 192.168.0.146
when i change peer-146 to 200 every thing is ok and i can call
from two side. but it is not good for me because 200 is the name of
extension and when i config asterisk systems, i don't know the name of
extensions, therefore i should use addresses not name of extensions.
do you know how i should define address of the other end in h323.conf
file? i define the address by host=192.168.0.146 but asterisk can not
find it? why?
On Mon, Apr 22, 2013 at 8:23 PM, Asghar Mohammad asghar...@gmail.com
wrote:
please post cli output for both calls.
On Mon, Apr 22, 2013 at 11:32 AM, s m sam.gh1...@gmail.com wrote:
hello everybody
i want to have sip connection between two asterisk systems (145 and
146). connection from 145 to 146 is ok but i can not call from 146
to
145.
this is h323.conf file in 145:
[peer146]
host=192.168.0.146
type=friend
context=from-trunk
[to-146]
type=peer
host=192.168.0.146
faststart=yes
tunneling=no
progress_audio=yes
disallow=all
allow=alaw
allow=ulaw
this is mu extensions.conf file in 145:
[from-trunk]
exten=_1.,1,Dial(SIP/to-231/1${EXTEN:1})
[line-231]
exten=_2.,1,Dial(H323/to-146/2${EXTEN:1})
i have this error: dropping call because extensions '100', 's' and
'i'
doesn't exists in context default.
if i change peer146 to general, every thing is ok and i can call
from two side. my question is: in h323 connection, is it a MUST to
have general context in h323.conf? if not, why i have this error
and
how i can solve it?
thanks in advance
sam
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Re: [asterisk-users] h323-sip: one way connection
flavor? i do not understand what you mean. please explain more.
thanks
On Wed, Apr 24, 2013 at 8:16 PM, Asghar Mohammad asghar...@gmail.comwrote:
what flavor of h323 you are using?
On Wed, Apr 24, 2013 at 8:50 AM, s m sam.gh1...@gmail.com wrote:
thanks Asghar,
i do it, but no thing happened:(
asterisk do not identify host line as ip address of the other end
On Tue, Apr 23, 2013 at 9:15 PM, Asghar Mohammad asghar...@gmail.comwrote:
try type=peer instead of friend.
On Tue, Apr 23, 2013 at 10:04 AM, s m sam.gh1...@gmail.com wrote:
i know what is the exactly problem. i enable debug for h323 and it
says:
could not find user by name 200 or address 192.168.0.146
when i change peer-146 to 200 every thing is ok and i can call from
two side. but it is not good for me because 200 is the name of extension
and when i config asterisk systems, i don't know the name of extensions,
therefore i should use addresses not name of extensions.
do you know how i should define address of the other end in h323.conf
file? i define the address by host=192.168.0.146 but asterisk can not
find it? why?
On Mon, Apr 22, 2013 at 8:23 PM, Asghar Mohammad
asghar...@gmail.comwrote:
please post cli output for both calls.
On Mon, Apr 22, 2013 at 11:32 AM, s m sam.gh1...@gmail.com wrote:
hello everybody
i want to have sip connection between two asterisk systems (145 and
146). connection from 145 to 146 is ok but i can not call from 146 to
145.
this is h323.conf file in 145:
[peer146]
host=192.168.0.146
type=friend
context=from-trunk
[to-146]
type=peer
host=192.168.0.146
faststart=yes
tunneling=no
progress_audio=yes
disallow=all
allow=alaw
allow=ulaw
this is mu extensions.conf file in 145:
[from-trunk]
exten=_1.,1,Dial(SIP/to-231/1${EXTEN:1})
[line-231]
exten=_2.,1,Dial(H323/to-146/2${EXTEN:1})
i have this error: dropping call because extensions '100', 's' and 'i'
doesn't exists in context default.
if i change peer146 to general, every thing is ok and i can call
from two side. my question is: in h323 connection, is it a MUST to
have general context in h323.conf? if not, why i have this error and
how i can solve it?
thanks in advance
sam
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Re: [asterisk-users] h323-sip: one way connection
thanks Asghar,
i do it, but no thing happened:(
asterisk do not identify host line as ip address of the other end
On Tue, Apr 23, 2013 at 9:15 PM, Asghar Mohammad asghar...@gmail.comwrote:
try type=peer instead of friend.
On Tue, Apr 23, 2013 at 10:04 AM, s m sam.gh1...@gmail.com wrote:
i know what is the exactly problem. i enable debug for h323 and it says:
could not find user by name 200 or address 192.168.0.146
when i change peer-146 to 200 every thing is ok and i can call from
two side. but it is not good for me because 200 is the name of extension
and when i config asterisk systems, i don't know the name of extensions,
therefore i should use addresses not name of extensions.
do you know how i should define address of the other end in h323.conf
file? i define the address by host=192.168.0.146 but asterisk can not
find it? why?
On Mon, Apr 22, 2013 at 8:23 PM, Asghar Mohammad asghar...@gmail.comwrote:
please post cli output for both calls.
On Mon, Apr 22, 2013 at 11:32 AM, s m sam.gh1...@gmail.com wrote:
hello everybody
i want to have sip connection between two asterisk systems (145 and
146). connection from 145 to 146 is ok but i can not call from 146 to
145.
this is h323.conf file in 145:
[peer146]
host=192.168.0.146
type=friend
context=from-trunk
[to-146]
type=peer
host=192.168.0.146
faststart=yes
tunneling=no
progress_audio=yes
disallow=all
allow=alaw
allow=ulaw
this is mu extensions.conf file in 145:
[from-trunk]
exten=_1.,1,Dial(SIP/to-231/1${EXTEN:1})
[line-231]
exten=_2.,1,Dial(H323/to-146/2${EXTEN:1})
i have this error: dropping call because extensions '100', 's' and 'i'
doesn't exists in context default.
if i change peer146 to general, every thing is ok and i can call
from two side. my question is: in h323 connection, is it a MUST to
have general context in h323.conf? if not, why i have this error and
how i can solve it?
thanks in advance
sam
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Re: [asterisk-users] h323-sip: one way connection
i know what is the exactly problem. i enable debug for h323 and it says:
could not find user by name 200 or address 192.168.0.146
when i change peer-146 to 200 every thing is ok and i can call from two
side. but it is not good for me because 200 is the name of extension and
when i config asterisk systems, i don't know the name of extensions,
therefore i should use addresses not name of extensions.
do you know how i should define address of the other end in h323.conf file?
i define the address by host=192.168.0.146 but asterisk can not find it?
why?
On Mon, Apr 22, 2013 at 8:23 PM, Asghar Mohammad asghar...@gmail.comwrote:
please post cli output for both calls.
On Mon, Apr 22, 2013 at 11:32 AM, s m sam.gh1...@gmail.com wrote:
hello everybody
i want to have sip connection between two asterisk systems (145 and
146). connection from 145 to 146 is ok but i can not call from 146 to
145.
this is h323.conf file in 145:
[peer146]
host=192.168.0.146
type=friend
context=from-trunk
[to-146]
type=peer
host=192.168.0.146
faststart=yes
tunneling=no
progress_audio=yes
disallow=all
allow=alaw
allow=ulaw
this is mu extensions.conf file in 145:
[from-trunk]
exten=_1.,1,Dial(SIP/to-231/1${EXTEN:1})
[line-231]
exten=_2.,1,Dial(H323/to-146/2${EXTEN:1})
i have this error: dropping call because extensions '100', 's' and 'i'
doesn't exists in context default.
if i change peer146 to general, every thing is ok and i can call
from two side. my question is: in h323 connection, is it a MUST to
have general context in h323.conf? if not, why i have this error and
how i can solve it?
thanks in advance
sam
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[asterisk-users] h323-sip: one way connection
hello everybody
i want to have sip connection between two asterisk systems (145 and
146). connection from 145 to 146 is ok but i can not call from 146 to
145.
this is h323.conf file in 145:
[peer146]
host=192.168.0.146
type=friend
context=from-trunk
[to-146]
type=peer
host=192.168.0.146
faststart=yes
tunneling=no
progress_audio=yes
disallow=all
allow=alaw
allow=ulaw
this is mu extensions.conf file in 145:
[from-trunk]
exten=_1.,1,Dial(SIP/to-231/1${EXTEN:1})
[line-231]
exten=_2.,1,Dial(H323/to-146/2${EXTEN:1})
i have this error: dropping call because extensions '100', 's' and 'i'
doesn't exists in context default.
if i change peer146 to general, every thing is ok and i can call
from two side. my question is: in h323 connection, is it a MUST to
have general context in h323.conf? if not, why i have this error and
how i can solve it?
thanks in advance
sam
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Re: [asterisk-users] Dropping call because extensions '200', 's' and 'i' doesn't exists
thanks guys, i solve my problem.
as Asghar said, i remove 2 and forget to add it again therefore
asterisk can not recognize extension 200 in extension.conf file.
this is my extension that works properly:
exten=_2.,1,Dial(SIP/to-231/1${EXTEN:2})
thanks every body for your attention.
Sam
On 4/13/13, Gertjan Baarda gertjan.baa...@gmail.com wrote:
Can you post both extensions.conf from both systems?
Sent from my iPhone
On 11 apr. 2013, at 14:51, s m sam.gh1...@gmail.com wrote:
this is my [from-trunk] extension:
[from-trunk]
exten=_2.,1,Dial(SIP/to-232/2${EXTEN:1})
and this is [to-231] in sip_additional.conf:
[to-232]
host=192.168.0.232
type=peer
qualify=yes
and 192.168.0.232 in the ip address of my freepbx.
On 4/11/13, A J Stiles asterisk_l...@earthshod.co.uk wrote:
On Thursday 11 April 2013, s m wrote:
when i call 100 from 200, every thing is ok and phone is ringing but
when i call 200 from 100, it says service unavailable.
i debug asterisk in my system 2 and see below message:
Dropping call because extensions '200', 's' and 'i' doesn't exists
in context [from-trunk]
OK. What do you have in the [from-trunk] context in your extensions.conf
?
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Re: [asterisk-users] Dropping call because extensions '200', 's' and 'i' doesn't exists
thanks Asghar, but it doesn't help. i have below error yet:(((
Dropping call because extensions '200', 's' and 'i' doesn't exists in
context [from-trunk]
i think that something is wring with my extensions in extensions.conf
but i don't know how to fix it.
please let me know if you have any other suggestion.
thanks
sam
On 4/11/13, Asghar Mohammad asghar...@gmail.com wrote:
hi,
try
exten= _2.,1,Dial(SIP/to-232/2${EXTEN:1})
Note space before underscore.
On Thu, Apr 11, 2013 at 2:50 PM, s m sam.gh1...@gmail.com wrote:
this is my [from-trunk] extension:
[from-trunk]
exten=_2.,1,Dial(SIP/to-232/2${EXTEN:1})
and this is [to-231] in sip_additional.conf:
[to-232]
host=192.168.0.232
type=peer
qualify=yes
and 192.168.0.232 in the ip address of my freepbx.
On 4/11/13, A J Stiles asterisk_l...@earthshod.co.uk wrote:
On Thursday 11 April 2013, s m wrote:
when i call 100 from 200, every thing is ok and phone is ringing but
when i call 200 from 100, it says service unavailable.
i debug asterisk in my system 2 and see below message:
Dropping call because extensions '200', 's' and 'i' doesn't exists
in context [from-trunk]
OK. What do you have in the [from-trunk] context in your
extensions.conf ?
--
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Answers come *after* questions.
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[asterisk-users] Dropping call because extensions '200', 's' and 'i' doesn't exists
hello all i,m newbie in asterisk and now want to sip and h323 connection. this is my scenario: phone(ext100)---freepbx---sip---system1---H323---system2---freepbx---phone(ext200) when i call 100 from 200, every thing is ok and phone is ringing but when i call 200 from 100, it says service unavailable. i debug asterisk in my system 2 and see below message: Dropping call because extensions '200', 's' and 'i' doesn't exists in context [from-trunk] i googled about this message and found that file extensions_mor_h323.conf should be included into /etc/asterisk/extensions_mor.conf. but i don't have any extensions_mor.conf file at all!!! is extensions_mor.conf really necessary to fix my problem?if yes, how i have connection in one way without this file? if no, how i can fix this problem? thanks in advance sam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropping call because extensions '200', 's' and 'i' doesn't exists
this is my [from-trunk] extension:
[from-trunk]
exten=_2.,1,Dial(SIP/to-232/2${EXTEN:1})
and this is [to-231] in sip_additional.conf:
[to-232]
host=192.168.0.232
type=peer
qualify=yes
and 192.168.0.232 in the ip address of my freepbx.
On 4/11/13, A J Stiles asterisk_l...@earthshod.co.uk wrote:
On Thursday 11 April 2013, s m wrote:
when i call 100 from 200, every thing is ok and phone is ringing but
when i call 200 from 100, it says service unavailable.
i debug asterisk in my system 2 and see below message:
Dropping call because extensions '200', 's' and 'i' doesn't exists
in context [from-trunk]
OK. What do you have in the [from-trunk] context in your extensions.conf ?
--
AJS
Answers come *after* questions.
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