Re: [Asterisk-Users] voip to voip bridge
Extracted from http://www.voip-info.org/wiki-Asterisk+cmd+Dial: ' When options /t/, /T, h, H, w, W or L (with multiple arguments) are applied, Asterisk will remain in the media path, even if /canreinvite=yes'' (a SIP channel option) has been specified.' Then how is it possible to limit a call without the L option ? Benoît Mérouze wrote: Hi, I've got some problems with bridged calls, the quality is extremely poor (more or less blanks or one way voice issues). But if I do a normal call with the same provider, there is no problem. Reinvite is enabled, but what are the parameters in the dial command that force asterisk to stay in the loop ? Are the H (to allow caller to hang up by dialing *) or L (to limit the call) parameters ones of them ? As an example, here is a Dial command I execute to bridge a call to a new one : SIP/kddi/0033172699611|30|HL(162:6:3) Thanks, Benoit [EMAIL PROTECTED] wrote: Hi, Check if reinvites are enabled, and that you don’t use any parameter in the dial command that forces asterisk to stay in the loop. Ohad *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Erick Baum *Sent:* Wednesday, June 14, 2006 5:00 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [Asterisk-Users] voip to voip bridge Has anyone had any good experiences with a voip to voip bridge... where you have an incoming call on a voip line which is redirected out another voip line to a regular phone line? Whenever we do this, the connected call is kinda lagged and the quality isn't always that great. It seems to me this is just a problem with the inherent delay in the voip connections. But I was wondering if there's any special configurations that could make the situation better? Erick -- Benoît Mérouze _._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._. Groupe IPercom - The VoIP Enabling Company - http://www.ipercom.com Ingénieur RD - courriel : [EMAIL PROTECTED] Network Software Developer - mailto: [EMAIL PROTECTED] Tél. / Phone : +33 1 7269 9611 ._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._ Siège Social 43, rue Fessart 92100 Boulogne Billancourt RCS NANTERRE B 440 345 528 - Capital social: 100 000 € CE COURRIEL COMME LES DOCUMENTS EVENTUELLEMENT ASSOCIES SONT CONFIDENTIELS COUVERTS PAR LE SECRET PROFESSIONNEL THIS E MAIL AND ANY DOCUMENT POSSIBLY ATTACHED ARE CONFIDENTIAL AND COVERED BY THE PROFESSIONAL SECRECY ._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._ Only two things are infinite, the universe and human stupidity, and I'm not sure about the former. Albert Einstein ___ --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] voip to voip bridge
Hi, I've got some problems with bridged calls, the quality is extremely poor (more or less blanks or one way voice issues). But if I do a normal call with the same provider, there is no problem. Reinvite is enabled, but what are the parameters in the dial command that force asterisk to stay in the loop ? Are the H (to allow caller to hang up by dialing *) or L (to limit the call) parameters ones of them ? As an example, here is a Dial command I execute to bridge a call to a new one : SIP/kddi/0033172699611|30|HL(162:6:3) Thanks, Benoit [EMAIL PROTECTED] wrote: Hi, Check if reinvites are enabled, and that you don’t use any parameter in the dial command that forces asterisk to stay in the loop. Ohad *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Erick Baum *Sent:* Wednesday, June 14, 2006 5:00 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [Asterisk-Users] voip to voip bridge Has anyone had any good experiences with a voip to voip bridge... where you have an incoming call on a voip line which is redirected out another voip line to a regular phone line? Whenever we do this, the connected call is kinda lagged and the quality isn't always that great. It seems to me this is just a problem with the inherent delay in the voip connections. But I was wondering if there's any special configurations that could make the situation better? Erick -- Benoît Mérouze _._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._. Groupe IPercom - The VoIP Enabling Company - http://www.ipercom.com Ingénieur RD - courriel : [EMAIL PROTECTED] Network Software Developer - mailto: [EMAIL PROTECTED] Tél. / Phone : +33 1 7269 9611 ._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._ Siège Social 43, rue Fessart 92100 Boulogne Billancourt RCS NANTERRE B 440 345 528 - Capital social: 100 000 € CE COURRIEL COMME LES DOCUMENTS EVENTUELLEMENT ASSOCIES SONT CONFIDENTIELS COUVERTS PAR LE SECRET PROFESSIONNEL THIS E MAIL AND ANY DOCUMENT POSSIBLY ATTACHED ARE CONFIDENTIAL AND COVERED BY THE PROFESSIONAL SECRECY ._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._._ Only two things are infinite, the universe and human stupidity, and I'm not sure about the former. Albert Einstein ___ --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] voip to voip bridge
Hi, Check if reinvites are enabled, and that you dont use any parameter in the dial command that forces asterisk to stay in the loop. Ohad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erick Baum Sent: Wednesday, June 14, 2006 5:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] voip to voip bridge Has anyone had any good experiences with a voip to voip bridge... where you have an incoming call on a voip line which is redirected out another voip line to a regular phone line? Whenever we do this, the connected call is kinda lagged and the quality isn't always that great. It seems to me this is just a problem with the inherent delay in the voip connections. But I was wondering if there's any special configurations that could make the situation better? Erick ___ --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
