[Asterisk-Users] Asterisk codec detection problem !
Hi , Here is my scenario : I have an 7940 Cisco IP phone which can handle g711ulaw,g711alaw and g729 codecs itself. A Quintum A800 which configured to work with g711ulaw and a Cisco Gateway which configured to accept g729 codecs only. The Quintum gateway is local gateway which connected to PSTN for Local PSTN calls and Cisco Gateway is on the internet which can handle international Calls. The problem is that when I configure my sip.conf like this on my asterisk : [Cisco-7940] type=friend secret= username=hamid host=dynamic disallow=all allow=ulaw alllow=g729 [quintum] type=peer context=from-pstn host=xxx.xxx.xxx.xxx callerid=From PSTN disallow=all allow=ulaw [Cisco-Int] type=peer host=xxx.xxx.xxx.xxx disallow=all allow=g729 the preffered codec for cisco-7940 will be ulaw and it will work OK for calls to PSTN ( I mean quintum ). but when I want to call international numbers ( I mean Cisco-Int ) it will warn that there is no g729 to ulaw translation available and will disconnect the call ( Cause I don't have chan_g729.so on my asterisk ) however my own Cisco-7940 can handle g729 codec itself and the asterisk should just passthrough ( make bridge ) to work with g729 codec. The same problem will happen for quintum if I change the allow=g729 position with allow=ulaw on cisco-7940 config. How can I solve this ? BTW I check the same scenario with OpenSER and everythings working like a charm without any problem. I mean my PSTN calls go through the Quintum with ulaw codec and the international calls go through the Cisco with g729 as codec without any problem. Regards Hamid Hashemi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pickup() h323
I did try it again without success. I did check the debug logs and there is nothing special there about any errors. Following the logs it says that the connection is established but no Voice and no Tone. here is my scenario : I have a SIP phone which make a SIP call to asterisk with G729 Codec. The Asterisk then make an H323 call to the external peer with G729 codec again and it should make bridge between these 2 calls ( 1 incomming and 1 outgoing ) I checked it with OH323 with the same scenario and it is working well. But for H323 I couldn't make the call. Any idea ? Hamid Hashemi wrote: Ok I will try it again and will let the list know about the result. Jeremy McNamara wrote: Hamid Hashemi wrote: Hi , Is your chan_h323 driver can support codec g729 or g723 in bridge mode ( I mean no transcoding just bridging ) ? I did try it without success however both chan_oh323 and chan_ooh323 can support it without any problem . Yes, chan_h323 will pass thru any codec as long as your configuration and the 'far-end' allows the proper codec. Also, you cannot have any Dial options (Ttr etc) and you cannot play any prompts. Just Dial,H323/your_location If you have trouble turn on H.323 debug and have debug on the console line in logger.conf. Jeremy McNamara ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards = / Seyyed Hamid Reza/WINDOWS FOR NOW !!/ / Hashemi Golpayegani / Linux for future , FreeBSD for ever / /Morva System Co. / - / / Network Administrator/ [EMAIL PROTECTED] , ICQ# : 42209876 / ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pickup() h323
Ok I will try it again and will let the list know about the result. Jeremy McNamara wrote: Hamid Hashemi wrote: Hi , Is your chan_h323 driver can support codec g729 or g723 in bridge mode ( I mean no transcoding just bridging ) ? I did try it without success however both chan_oh323 and chan_ooh323 can support it without any problem . Yes, chan_h323 will pass thru any codec as long as your configuration and the 'far-end' allows the proper codec. Also, you cannot have any Dial options (Ttr etc) and you cannot play any prompts. Just Dial,H323/your_location If you have trouble turn on H.323 debug and have debug on the console line in logger.conf. Jeremy McNamara ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards = / Seyyed Hamid Reza/WINDOWS FOR NOW !!/ / Hashemi Golpayegani / Linux for future , FreeBSD for ever / /Morva System Co. / - / / Network Administrator/ [EMAIL PROTECTED] , ICQ# : 42209876 / ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pickup() h323
Hi , Is your chan_h323 driver can support codec g729 or g723 in bridge mode ( I mean no transcoding just bridging ) ? I did try it without success however both chan_oh323 and chan_ooh323 can support it without any problem . _Hamid Jeremy McNamara wrote: Pavel Jezek wrote: Hello Jeremy, do you think, that adding features to original h323 channel is perspective? is still maintained or will be replaced eg. with ooh323 (from asterisk add-ons)? anyway I'm currently using original h323, it working prety fine for me (with ooh323/oh323 I had problem with callerid between h323-asterisk)... chan_h323 is very much supported, just nobody has bothered to give me any valid information on what needs to be fixed. I have totally removed H.323 from my operation, so I no longer utilize chan_h323 for anything. Thus it is now up to the community to report issues they find. Digium paid for ooh323, for whatever reasons that is beyond me, but it has proven to be no better than any H.323 channel driver, so I hope they got their money back. Jeremy McNamara ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SER inside of Asterisk is SCALABLE ?
Hi, I read somewhere if you need an scalable Gateway for huge number of agent it is better to use SER ( OpenSER ) inside of your asterisk. My question is that is it really true ? and why ? Also is there any way to integrate SER with asterisk in this way that the SER itself won't route any calls to the SIP clients which register with their own custom numbers but check it with asterisk route rules and then route the calls. -- Regards Hamid Hashemi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
