Fw: Re: Re: [Asterisk-Users] inter Asterisk trunking IAX /IAX2
I originally wanted to answer with something ... tzarit and kevit Readed probably before you invent rapid biz. I am asking to share any info /experience not your high spirit. Thanks for less trivial answer, G On Wed, Oct 05, 2005 at 12:44:27PM -0700, Thameem Ansari wrote: I am using the inter asterisk trunking and the article in voip-info.orghttp://voip-info.orgwill not work. I originally wanted to answer with something like: is it on strike? . If you want to get help here, please provide useful information to help with. Which article exactly? Could you please give the full URL? On what system(s) did you try it? What exactly did you do? What did happen? (error messages, CLI traces, etc.) How do you call from one Asterisk to another? What happens when you try to call? Could you provide the relevant config files? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] inter Asterisk trunking IAX /IAX2
Geo [EMAIL PROTECTED] wrote: Anyone using inter Asterisk trunking IAX /IAX2 ? No - you're the first to think of that. Congratulations. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] inter Asterisk trunking IAX /IAX2
On Wed, Oct 05, 2005 at 03:51:38PM -0800, Geo wrote: Hi, Anyone using inter Asterisk trunking IAX /IAX2 ? Thanks, Not sure about IAX (1), but IAX2 is widely used. Before asking trivial questions you probably should take the time reading about it in http://voip-info/wiki-Asterisk and similar places, though. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] inter Asterisk trunking IAX /IAX2
I am using the inter asterisk trunking and the article in voip-info.org will not work. On 10/5/05, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Oct 05, 2005 at 03:51:38PM -0800, Geo wrote: Hi, Anyone using inter Asterisk trunking IAX /IAX2 ? Thanks,Not sure about IAX (1), but IAX2 is widely used. Before asking trivial questions you probably should take the time reading about it inhttp://voip-info/wiki-Asterisk and similar places, though.--Tzafrir Cohen | [EMAIL PROTECTED] | VIM ishttp://tzafrir.org.il | | a Mutt's[EMAIL PROTECTED] | |bestICQ# 16849755 | | friend___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] inter Asterisk trunking IAX /IAX2
On Wed, Oct 05, 2005 at 12:44:27PM -0700, Thameem Ansari wrote: I am using the inter asterisk trunking and the article in voip-info.orghttp://voip-info.orgwill not work. I originally wanted to answer with something like: is it on strike? . If you want to get help here, please provide useful information to help with. Which article exactly? Could you please give the full URL? On what system(s) did you try it? What exactly did you do? What did happen? (error messages, CLI traces, etc.) How do you call from one Asterisk to another? What happens when you try to call? Could you provide the relevant config files? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] inter-asterisk meetme
Why do you want to do that? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zen Kato Sent: Wednesday, August 03, 2005 5:41 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] inter-asterisk meetme Hi, If there are 5 asterisk servers on the local net and each server runs meetme, eg. 3311,3321,3331,3341,3351 respectively. Can I connect these 5 meetme conferences to one meetme using IAX2? Regards, Zen ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.9/62 - Release Date: 8/2/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.9/62 - Release Date: 8/2/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] inter-asterisk meetme
William Boehlke wrote : Why do you want to do that? 100 sip users are connected to each CPU(P4 3.0MHz)s, then I would like to broadcast from one sip phone to 500 sip users. If I have 5 microphones in front of me, I can talk to 5 microphones, then 500 users can listen(one-way mode) simultaneously. But that is not elegant. -- Zen -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zen Kato Sent: Wednesday, August 03, 2005 5:41 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] inter-asterisk meetme Hi, If there are 5 asterisk servers on the local net and each server runs meetme, eg. 3311,3321,3331,3341,3351 respectively. Can I connect these 5 meetme conferences to one meetme using IAX2? Regards, Zen ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.9/62 - Release Date: 8/2/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.9/62 - Release Date: 8/2/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] inter-asterisk meetme
I haven't played with it but I think you want to stream using ices. - Original Message - From: Zen Kato [EMAIL PROTECTED] To: asterisk-users@lists.digium.com; [EMAIL PROTECTED] Sent: Wednesday, August 03, 2005 6:20 PM Subject: Re: [Asterisk-Users] inter-asterisk meetme William Boehlke wrote : Why do you want to do that? 100 sip users are connected to each CPU(P4 3.0MHz)s, then I would like to broadcast from one sip phone to 500 sip users. If I have 5 microphones in front of me, I can talk to 5 microphones, then 500 users can listen(one-way mode) simultaneously. But that is not elegant. -- Zen -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zen Kato Sent: Wednesday, August 03, 2005 5:41 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] inter-asterisk meetme Hi, If there are 5 asterisk servers on the local net and each server runs meetme, eg. 3311,3321,3331,3341,3351 respectively. Can I connect these 5 meetme conferences to one meetme using IAX2? Regards, Zen ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.9/62 - Release Date: 8/2/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.9/62 - Release Date: 8/2/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] inter asterisk
I am trying to forward calls to another * server with IAX Here is What I want to Do 1- Call SERVER1, let say at 51412345678 2- SERVER1 should transfer the call to SERVER2 in a remote location 3- SERVER2 Receive the call and transfer it to the PSTN number. I have one X100P card on each machine. What is happening is that when the remote party picks up the phone, all he can hear is a weird sound. CONFIGS: SERVER1: zaptel.conf - ~ [channels] ~ language=fr ~ context=montréal ~ signalling=fxs_ks ~ usercallerid=yes ~ callwaiting=yes ~ threewaycalling=yes ~ transfer=yes ~ cancellforward=yes ~ echocancel=yes ~ echocancelwhenbridged=yes ~ echotraining=yes ~ relaxdtmf=yes ~ busydetect=yes ~ busycount=4 ~ callprogress=yes ~ group=1 ~ channel=1 -- (same for SERVER2) IAX.conf ~ [general] ~ bindport=4569 ~ delayreject=yes ~ language=fr ~ allow=all ~ jutterbuffer=no ~ register = username:[EMAIL PROTECTED] ~ tos=lowdelay ~ autokill=yes ~ ~ [quebec] ~ type=friends ~ username = username ~ password=password ~ context=montréal ~ host=Dynamic ~ secret = password ~ disallow = all ~ allow=ulaw ~ allow=gsm extensions.conf --(Same for SERVER2 but no registration) ~ [general] ~ static=yes ~ writeprotect=yes ~ autofallthrough=yes ~ [montréal] ~ exten=s,1,Answer ~ exten=s,2,Playback(message-transfer) ~ exten=s,3,Dial(IAX2/username:[EMAIL PROTECTED]/[EMAIL PROTECTED]) ; always the same number ~ exten=s,4,Hangup My remote server receive the call, answer the line and then Dial(ZAP/1/51412345678). So far so good. But when 51412345678 pickup the phone, all she can hear is a weird sound. What am I doing wrong ? Difficult to tell without some feedback from the CLI. If you actually copy/pasted the above config statements, I'm assuming you manually added all those ~ at the front of each line. If they are actually in your config, get rid of them. The statement jutterbuffer=no should be jitterbuffer=no. One thing you might try to at least eliminate possible problems is to change iax.conf to disallow=all and allow=gsm only. Get rid of the allow=ulaw and do another test. Might as well add trunk=no to this link as well. (Must stop and restart * after making these type changes.) You might try 'iax2 debug' from the CLI on both machines and look at the detail to see if you can spot any conflicts or problems. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] inter asterisk
One thing I do on remote sites is set up a soft phone so I can call myself, this proves out the link and quality before anything else. DIAX id good for this as you can connect to multiple sites, also good to see if you have problems before anyone else calls you to say there is a problem. It also helps in cases like this, if your return quality is good then the possible fault lies with the ZAP interface. Process of elimination, works for me every time. Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ousmane Doukara Sent: 06 February 2005 08:46 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] inter asterisk Hi, I am trying to forward calls to another * server with IAX Here is What I want to Do 1- Call SERVER1, let say at 51412345678 2- SERVER1 should transfer the call to SERVER2 in a remote location 3- SERVER2 Receive the call and transfer it to the PSTN number. I have one X100P card on each machine. What is happening is that when the remote party picks up the phone, all he can hear is a weird sound. CONFIGS: SERVER1: zaptel.conf - ~ [channels] ~ language=fr ~ context=montréal ~ signalling=fxs_ks ~ usercallerid=yes ~ callwaiting=yes ~ threewaycalling=yes ~ transfer=yes ~ cancellforward=yes ~ echocancel=yes ~ echocancelwhenbridged=yes ~ echotraining=yes ~ relaxdtmf=yes ~ busydetect=yes ~ busycount=4 ~ callprogress=yes ~ group=1 ~ channel=1 -- (same for SERVER2) IAX.conf ~ [general] ~ bindport=4569 ~ delayreject=yes ~ language=fr ~ allow=all ~ jutterbuffer=no ~ register = username:[EMAIL PROTECTED] ~ tos=lowdelay ~ autokill=yes ~ ~ [quebec] ~ type=friends ~ username = username ~ password=password ~ context=montréal ~ host=Dynamic ~ secret = password ~ disallow = all ~ allow=ulaw ~ allow=gsm extensions.conf --(Same for SERVER2 but no registration) ~ [general] ~ static=yes ~ writeprotect=yes ~ autofallthrough=yes ~ [montréal] ~ exten=s,1,Answer ~ exten=s,2,Playback(message-transfer) ~ exten=s,3,Dial(IAX2/username:[EMAIL PROTECTED]/[EMAIL PROTECTED] al) ; always the same number ~ exten=s,4,Hangup My remote server receive the call, answer the line and then Dial(ZAP/1/51412345678). So far so good. But when 51412345678 pickup the phone, all she can hear is a weird sound. What am I doing wrong ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] inter asterisk
I think it has to do with my ZAP interface. Before my DIAL(ZAP/1/51412345678) I have a Playback(message-transfert) which play nicely. As soon as the ZAP start ringing the PSTN phone, i have that helicopter sound. - Original Message - From: David J Carter [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, February 06, 2005 7:18 AM Subject: RE: [Asterisk-Users] inter asterisk One thing I do on remote sites is set up a soft phone so I can call myself, this proves out the link and quality before anything else. DIAX id good for this as you can connect to multiple sites, also good to see if you have problems before anyone else calls you to say there is a problem. It also helps in cases like this, if your return quality is good then the possible fault lies with the ZAP interface. Process of elimination, works for me every time. Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ousmane Doukara Sent: 06 February 2005 08:46 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] inter asterisk Hi, I am trying to forward calls to another * server with IAX Here is What I want to Do 1- Call SERVER1, let say at 51412345678 2- SERVER1 should transfer the call to SERVER2 in a remote location 3- SERVER2 Receive the call and transfer it to the PSTN number. I have one X100P card on each machine. What is happening is that when the remote party picks up the phone, all he can hear is a weird sound. CONFIGS: SERVER1: zaptel.conf - ~ [channels] ~ language=fr ~ context=montréal ~ signalling=fxs_ks ~ usercallerid=yes ~ callwaiting=yes ~ threewaycalling=yes ~ transfer=yes ~ cancellforward=yes ~ echocancel=yes ~ echocancelwhenbridged=yes ~ echotraining=yes ~ relaxdtmf=yes ~ busydetect=yes ~ busycount=4 ~ callprogress=yes ~ group=1 ~ channel=1 -- (same for SERVER2) IAX.conf ~ [general] ~ bindport=4569 ~ delayreject=yes ~ language=fr ~ allow=all ~ jutterbuffer=no ~ register = username:[EMAIL PROTECTED] ~ tos=lowdelay ~ autokill=yes ~ ~ [quebec] ~ type=friends ~ username = username ~ password=password ~ context=montréal ~ host=Dynamic ~ secret = password ~ disallow = all ~ allow=ulaw ~ allow=gsm extensions.conf --(Same for SERVER2 but no registration) ~ [general] ~ static=yes ~ writeprotect=yes ~ autofallthrough=yes ~ [montréal] ~ exten=s,1,Answer ~ exten=s,2,Playback(message-transfer) ~ exten=s,3,Dial(IAX2/username:[EMAIL PROTECTED]/[EMAIL PROTECTED] al) ; always the same number ~ exten=s,4,Hangup My remote server receive the call, answer the line and then Dial(ZAP/1/51412345678). So far so good. But when 51412345678 pickup the phone, all she can hear is a weird sound. What am I doing wrong ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Inter-Asterisk Exchange
Bryan: I would like to setup 2 Asterisk boxes. One would be located in our office behind the firewall and hooked up to our analog lines. The other would be located in a remote datacenter and used for our remote employees to connect to. I would like to be able to accept calls on the Office Asterisk server and route them to the Datacenter Asterisk server. Is this possible? Trivial. We're doing just that for a client over an RF link to get phone service to a building which would otherwise have to wait months for land lines. Depending on what you are planning to do in the datacenter you could just put SIP phones/ATAs there rather than a full Asterisk server but that would require some care in configuring your firewall. George Pajari netVOICE communications www.netvoice.ca www.ip-centrex.ca ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Inter-Asterisk Exchange
On Fri, 2004-07-02 at 15:49, Bryan Brannigan wrote: I would like to setup 2 Asterisk boxes. One would be located in our office behind the firewall and hooked up to our analog lines. The other would be located in a remote datacenter and used for our remote employees to connect to. I would like to be able to accept calls on the Office Asterisk server and route them to the Datacenter Asterisk server. Is this possible? Yes. Kanwar Systems Aligned Inc. www.systemsaligned.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Inter-Asterisk Exchange
Depending on what you are planning to do in the datacenter you could just put SIP phones/ATAs there rather than a full Asterisk server but that would require some care in configuring your firewall. Actually the users are will be remote to the datacenter. The IPs in our office are dynamic so I imagine that would be an issue to just hosting the box there. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Inter-Asterisk Exchange
On Fri, 2004-07-02 at 16:22, Bryan Brannigan wrote: Depending on what you are planning to do in the datacenter you could just put SIP phones/ATAs there rather than a full Asterisk server but that would require some care in configuring your firewall. Actually the users are will be remote to the datacenter. The IPs in our office are dynamic so I imagine that would be an issue to just hosting the box there. To prevent yourself from losing hair, one side, preferably the main office, should have static IPs. Then the remote users could connect to the main office through a VPN, and viola, they're sitting on the same network as the main office. But, the remote users wouldn't need an Asterisk server. It wouldn't matter if their IPs changed because the VPN could be configured at either end to take that into account. Kanwar Systems Aligned Inc. www.systemsaligned.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
