RE: [Asterisk-Users] International area codes (incl. mobile)
I can send a list, mobile is not complete but it has a lot of numbers... -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de PHP Mechanic Enviado el: Viernes, 07 de Enero de 2005 11:57 a.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] International area codes (incl. mobile) Hello everybody, does anybody knows from where I can get an list of international area codes incl. the mobile numbers? Have you tried google ? http://www.google.com.au/search?hl=enq=international+dialing+codes ___ 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: 7.0.300 / Virus Database: 265.6.9 - Release Date: 2005-01-06 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.9 - Release Date: 2005-01-06 ___ 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] OT: List of VoIP providers?
Voipproviderlist.com -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Jeromie Reeves Enviado el: Martes, 04 de Enero de 2005 03:30 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [Asterisk-Users] OT: List of VoIP providers? I have been looking around for VoIP providers but have not found a good listing. Is there no yellow pages for VoIP providers? Google mostly returns services like Vonage, Packet8, NuFone, ect. None seam to be very reseller friendly and none offer LNP or local DID's for my area. Anyone know of a list (even a partial one) Jeromie Reeves ___ 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 --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.803 / Virus Database: 546 - Release Date: 2004-11-30 --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.803 / Virus Database: 546 - Release Date: 2004-11-30 ___ 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] h.323 Type=User
is h323 per user based working??? I have setup this: [User1]type=userhost=xx.xx.xx.xx context=international incominglimit=30 But all calls from xx.xx.xx.xx are not routed to context international, it is working? I am using chan_h323 Thanks!! Sebastian Nocetti. --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.803 / Virus Database: 546 - Release Date: 2004-11-30 ___ 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] h.323 Type=User
Thanks !! I will try!! -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Soren Rathje Enviado el: Martes, 21 de Diciembre de 2004 02:30 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] h.323 Type=User Sebastian Nocetti wrote: is h323 per user based working??? I have setup this: [User1] type=user host=xx.xx.xx.xx context=international incominglimit=30 But all calls from xx.xx.xx.xx are not routed to context international, it is working? I am using chan_h323 I'm using current CVS 21-12-04 and found that my Siemens OptiPoint 300a w/ h323 version 2.5.32 (and no GK) have some issues with openh323, it sends the h323-id but chan_h323 is not able to attach a context to it except for the default context... I found that by adding userbyalias = no to h323.conf it now associate device/context by IP address and not Name... It works for me but it is apparent to me that the h323 stack in the phone is pure crap.. :-) /Soren ___ 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 --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.803 / Virus Database: 546 - Release Date: 2004-11-30 --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.803 / Virus Database: 546 - Release Date: 2004-11-30 ___ 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] h.323 Type=User
Now it is working... Thanks! -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Soren Rathje Enviado el: Martes, 21 de Diciembre de 2004 02:30 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] h.323 Type=User Sebastian Nocetti wrote: is h323 per user based working??? I have setup this: [User1] type=user host=xx.xx.xx.xx context=international incominglimit=30 But all calls from xx.xx.xx.xx are not routed to context international, it is working? I am using chan_h323 I'm using current CVS 21-12-04 and found that my Siemens OptiPoint 300a w/ h323 version 2.5.32 (and no GK) have some issues with openh323, it sends the h323-id but chan_h323 is not able to attach a context to it except for the default context... I found that by adding userbyalias = no to h323.conf it now associate device/context by IP address and not Name... It works for me but it is apparent to me that the h323 stack in the phone is pure crap.. :-) /Soren ___ 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 --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.803 / Virus Database: 546 - Release Date: 2004-11-30 --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.803 / Virus Database: 546 - Release Date: 2004-11-30 ___ 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] Asterisk + AS5300
I am doing that actually, terminating calls via SIP on a Cisco AS5300, and it is working good. De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de FranciscoEnviado el: Miércoles, 01 de Diciembre de 2004 10:43 a.m.Para: [EMAIL PROTECTED]Asunto: [Asterisk-Users] Asterisk + AS5300 Is it possible to terminate calls via SIP on a Cisco AS5300? Did anyone do it?How? Do i need an special IOS version? Ive beentrying to compile the OpenH323 channel for the last month, but errors still happens. Thanks in advance. ---Checked by AVG anti-virus system (http://www.grisoft.com).Version: 6.0.801 / Virus Database: 544 - Release Date: 2004-11-24 --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.801 / Virus Database: 544 - Release Date: 2004-11-24 ___ 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] Asterisk + AS5300
ok, this is my config. sip.conf [gw-as5300] type=friendinsecure=yeshost=xxx.xxx.xx.xx disallow=allallow=g729allow=ulawcanreinvite=noreinvite=nodtmfmode=rfc2833 extensions.conf exten = _.,1,Dial(SIP/[EMAIL PROTECTED]) exten = _.,2,Congestion 5300 dialpeer, by default cisco creates a VOIP dialpeer for incoming. or you can set your own. I have a E1 R2 connected to my cisco as5300 to terminate calls to PSTN. dial-peer voice 1 pots destination-pattern 100#T port 2:0 ! De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de FranciscoEnviado el: Miércoles, 01 de Diciembre de 2004 11:05 a.m.Para: Asterisk Users Mailing List - Non-Commercial DiscussionAsunto: Re: [Asterisk-Users] Asterisk + AS5300 Can you post a sample of your configuration? (sip.conf, extensions.conf and as5300 dial-peers) Thanks! boch.- - Original Message - From: Sebastian Nocetti To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Wednesday, December 01, 2004 10:54 AM Subject: RE: [Asterisk-Users] Asterisk + AS5300 I am doing that actually, terminating calls via SIP on a Cisco AS5300, and it is working good. De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de FranciscoEnviado el: Miércoles, 01 de Diciembre de 2004 10:43 a.m.Para: [EMAIL PROTECTED]Asunto: [Asterisk-Users] Asterisk + AS5300 Is it possible to terminate calls via SIP on a Cisco AS5300? Did anyone do it?How? Do i need an special IOS version? Ive beentrying to compile the OpenH323 channel for the last month, but errors still happens. Thanks in advance. ---Checked by AVG anti-virus system (http://www.grisoft.com).Version: 6.0.801 / Virus Database: 544 - Release Date: 2004-11-24 ---Checked by AVG anti-virus system (http://www.grisoft.com).Version: 6.0.801 / Virus Database: 544 - Release Date: 2004-11-24 ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---Checked by AVG anti-virus system (http://www.grisoft.com).Version: 6.0.801 / Virus Database: 544 - Release Date: 2004-11-24 --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.801 / Virus Database: 544 - Release Date: 2004-11-24 ___ 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] cisco dial-peer voip
I think you CAN'T DO VOIP-VOIP into CISCO Equipment, it have to be POTS-VOIP or viceversa. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Brian Wilkins Enviado el: Martes, 30 de Noviembre de 2004 05:57 a.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] cisco dial-peer voip Why not just have the Asterisk server act as a SIP/H323 gateway instead of the cisco router? You can then send incoming calls to registered Asterisk users via the cisco router and outgoing calls from Asterisk users to the PSTN via the cisco router. You can still use your same config below, but send the VoIP sessions through Asterisk and let it parse out where the calls need to go and send it to the cisco if you want to terminate traffic. On Tuesday 30 November 2004 01:35 pm, Jan Baggen wrote: I have 2610 XM with 1 Fastethernet and VIC2-2BRI. Dialin and dialout over pots is ok. Also inbound pots calls get redirected to Asterisk y.y.y.y So far so good. But I want to setup VOIP sessions with local carrier. I added dial-peer 40 for this. Session target x.x.x.x But calls will always get routed to the pstn peers 50 and 60. Peer x.x.x.x is never contacted or tried. My situation: PSTN - CISCO - ASTERISK OK ASTERISK - CISCO - PSTN OK ASTERISK - CISCO - VOIP NOT OK (only needs outbound calls) SIP01#sh dial-peer voice summary dial-peer hunt 0 TAGTYPE MIN OPER PREFIXDEST-PATTERN FER THRU SESS-TARGET STAT PORT 10 pots up up0 down 1/0/0 20 pots up up0 down 1/0/1 30 voip up up 2012345.. 0 syst ipv4:y.y.y.y:5060 40 voip up up .+ 0 syst ipv4:x.x.x.x:5060 50 pots up up .+ 5 up 1/0/0 60 pots up up .+ 5 up 1/0/1 dial-peer voice 10 pots description INBOUND CALLS PSTN BRI0 incoming called-number 2012345.. no digit-strip direct-inward-dial port 1/0/0 ! dial-peer voice 20 pots description INBOUND CALLS PSTN BRI1 incoming called-number 2012345.. no digit-strip direct-inward-dial port 1/0/1 ! dial-peer voice 30 voip description INBOUND CALLS VOIP ASTERISK destination-pattern 2051860.. session protocol sipv2 session target ipv4:y.y.y.y:5060 session transport udp dtmf-relay sip-notify codec g711alaw no vad ! dial-peer voice 40 voip description OUTBOUND CALLS VOIP CARRIER destination-pattern .+ session protocol sipv2 session target ipv4:x.x.x.x:5060 session transport tcp dtmf-relay sip-notify codec g711alaw no vad ! dial-peer voice 50 pots tone ringback alert-no-PI description OUTBOUND CALLS PSTN BRI0 preference 5 destination-pattern .+ no digit-strip port 1/0/0 ! dial-peer voice 60 pots tone ringback alert-no-PI description OUTBOUND CALLS PSTN BRI1 preference 5 destination-pattern .+ no digit-strip port 1/0/1 ___ 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 -- Brian Wilkins Software Engineer [EMAIL PROTECTED] Heritage Communications Corporation Melbourne, FL USA 32935 321.308.4000 x33 http://www.hcc.net ___ 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 --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.801 / Virus Database: 544 - Release Date: 2004-11-24 --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.801 / Virus Database: 544 - Release Date: 2004-11-24 ___ 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] ATA186 V2.15.ms
Check what IOS ata have installed... Because by default it does not comes with H.323 - SIP IOS... If you want I can send you both ios... Contact me at: [EMAIL PROTECTED] -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Rodney Acosta Coya Enviado el: Martes, 23 de Noviembre de 2004 01:54 p.m. Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Asunto: [Asterisk-Users] ATA186 V2.15.ms Importancia: Alta Hi I have a brand new ATA186 with the following firmware: Version: v2.15.ms ata186 (Build 020919a) I have been through the archives about how to configure it, but my colorful configuration web page does not have the same fields that people say I need to adjust. Even the examples on Cisco's web site don;t match. For example, I don't have the GtkOrProxy field, which is an important one. can some bady help me ??? thanks in advance Rodney Acosta Coya. ___ 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 --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.791 / Virus Database: 535 - Release Date: 2004-11-08 --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.791 / Virus Database: 535 - Release Date: 2004-11-08 ___ 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
[Asterisk-Users] H.323 Status
Hello all, somebody can tell me how h.323 status is? it is working OK?... it has implemented faststart and tunneling per peer based?... thanks a lot!! Sebastian from Argentina. --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.791 / Virus Database: 535 - Release Date: 2004-11-08 ___ 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] Asterisk as PSTN gateway
Asterisk works ok, but it have a lot of errors... 1st: It ever handle audio packet, and you cant do for exacmple only SIGNALLING 2st: It cant handle more than 20 channels simultaneous ... I tested it. 3st: It does not have fully Radius support.- -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Bill Hamlin Enviado el: Viernes, 24 de Septiembre de 2004 10:38 a.m. Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] Asterisk as PSTN gateway I've been asked to recommend a solution for a one-E1-port PSTN gateway supporting SIP. I've never set up a Cisco 5300 or equivalent, but I know they work. I use the Asterisk software in a couple of places and would like to use the E100P. My question is whether anyone out there has any installations using this and what their opinion is about it (does it work? how's the audio quality? and so on). Thanks, Bill Hamlin ___ 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 --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.744 / Virus Database: 496 - Release Date: 2004-08-24 --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.744 / Virus Database: 496 - Release Date: 2004-08-24 ___ 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] SIP termination in Brazil
I am interested too in termination using SIP to brazil, we need h.323 too... Can you contact me? Thanks Sebastian. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Daniel Bichara Enviado el: Martes, 21 de Septiembre de 2004 11:06 a.m. Para: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] SIP termination in Brazil Olá Julio, Também oferecemos IAX2. Daniel Julio Arruda wrote: Daniel Bichara wrote: Hi Han, Our company can offer you a SIP termination in Brazil up and running. Daniel IAX2 Termination ? I'm looking for some in Campinas, Sao Paulo and Rio de Janeiro. Johannes van Hulst wrote: Is there an up and running provider of SIP termination in Brazil? I know that there are some people building on a SIP termination solution. But who as it up and running ? ___ 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 ___ 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 --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.744 / Virus Database: 496 - Release Date: 2004-08-24 --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.744 / Virus Database: 496 - Release Date: 2004-08-24 ___ 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
[Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS
hello all, does anyone has experiencie using asterisk with a digium CARD using G729 managing 120 concurrent calls with SIP and/or H323??? I wanna know ifAsterisk is stable doing thisbecause we wanna implement it in some locations!! Thanks All!! Sebastian.
RE: [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS
E1's, only G729 and from SIP to E1 or from E1 to SIP De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de mattfEnviado el: Viernes, 06 de Agosto de 2004 03:44 p.m.Para: '[EMAIL PROTECTED]'Asunto: RE: [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS Will you have E1s? will you restrict users to 729 or will you allow other codecs? will most calls be from SIP to SIP? or SIP to E1 lines? MATT--- -Original Message-From: Sebastian Nocetti [mailto:[EMAIL PROTECTED]Sent: Friday, August 06, 2004 12:53 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS hello all, does anyone has experiencie using asterisk with a digium CARD using G729 managing 120 concurrent calls with SIP and/or H323??? I wanna know ifAsterisk is stable doing thisbecause we wanna implement it in some locations!! Thanks All!! Sebastian.
RE: [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS
Mm, that's bad, wow... Well, I will see howto implement that... Thanks for you comments Aaa... My macbhine is a DUAL XEON 3.4 with 2GB memory Is a HP PROLIANT. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de mattf Enviado el: Viernes, 06 de Agosto de 2004 04:07 p.m. Para: '[EMAIL PROTECTED]' Asunto: RE: [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS Nope, you won't be able to build a server fast enough to handle the transcoding. At the very most we've handled 60 concurrent SIP to T1 conversations on a Dual Athlon MP 2800+ system before it crashed, and I've never heard of anyone having more than 90 concurrent SIP to Zap channels running (and that was in a lab envorinment). If you want to use Asterisk you should look into multiple, fast asterisk servers handling 50 concurrent calls at the most each. MATT--- -Original Message- From: Sebastian Nocetti [mailto:[EMAIL PROTECTED] Sent: Friday, August 06, 2004 2:51 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS E1's, only G729 and from SIP to E1 or from E1 to SIP De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de mattf Enviado el: Viernes, 06 de Agosto de 2004 03:44 p.m. Para: '[EMAIL PROTECTED]' Asunto: RE: [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS Will you have E1s? will you restrict users to 729 or will you allow other codecs? will most calls be from SIP to SIP? or SIP to E1 lines? MATT--- -Original Message- From: Sebastian Nocetti [mailto:[EMAIL PROTECTED] Sent: Friday, August 06, 2004 12:53 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS hello all, does anyone has experiencie using asterisk with a digium CARD using G729 managing 120 concurrent calls with SIP and/or H323??? I wanna know if Asterisk is stable doing thisbecause we wanna implement it in some locations!! Thanks All!! Sebastian. ___ 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 ___ 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] H323 Call Dropping
Can you try with g711 to see if all is going ok? -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Asterisk . Enviado el: Jueves, 05 de Agosto de 2004 10:10 a.m. Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] H323 Call Dropping Hello, --- Sebastian Nocetti [EMAIL PROTECTED] wrote: Dial(h323/h323:[EMAIL PROTECTED]) I think problem is in this line... Dial(h323/[EMAIL PROTECTED]) That's not the correct way? Thanks for the reply. I tried both, also without mentioning the gatekeeper_ip in the dialplan. All the 3 methods were sending calls to the gatekeeper, then the calls were dropped. I traced the h323 traffic, and here are the excerpts of the trace, but why is it saying call EndedByNoBandWidth?! What am i doing wrong? (The codec is G723) 3:35.825 H225 Caller:81007e8 h323.cxx(1591) H225 Reading PDUs: callRef=17263 3:35.858GkMonitor:812dc70 h323.cxx(4406) H323 Bandwidth used: 0 3:35.859GkMonitor:812dc70 h323pdu.cxx(494) Trans Sending PDU: infoRequestResponse 57494 3:35.877 H225 Caller:81007e8 h323pdu.cxx(494) H225 Receiving PDU: releaseComplete 3:35.877 H225 Caller:81007e8 h323.cxx(1643) H225 Handling PDU: ReleaseComplete callRef=17263 3:35.878 H225 Caller:81007e8 h323neg.cxx(334) H245 Stopping MasterSlaveDetermination: state=Outgoing 3:35.878 H225 Caller:81007e8 h323neg.cxx(561) H245 Stopping TerminalCapabilitySet: state=InProgress 3:35.878 H225 Caller:81007e8 h323.cxx(1903) H225Set protocol version to 2 and implying H.245 version 3 3:35.878 H225 Caller:81007e8 h323ep.cxx(1681) H323 Clearing connection ip$localhost/17263 reason=EndedByNoBandwidth 3:35.879 H225 Caller:81007e8 h323.cxx(1427) H323Call end reason for ip$localhost/17263 set to EndedByNoBandwidth 3:35.879 H225 Caller:81007e8 h323.cxx(1445) H225 Sending release complete PDU: callRef=17263 3:35.880 H225 Caller:81007e8 h323pdu.cxx(494) H245 Sending PDU: command endSessionCommand 3:35.881 H225 Caller:81007e8 h323.cxx(3109) H245 Write PDU fail: no control channel. 3:35.881 H225 Caller:81007e8 h323pdu.cxx(494) H225 Sending PDU: releaseComplete 3:35.882 H323 Cleaner h323ep.cxx(1739) H323 Cleaning up connections 3:35.882 H323 Cleaner h323.cxx(1482) H323 Connection ip$localhost/17263 closing: connectionState=AwaitingSignalConnect 3:35.882 H323 Cleaner transports.cxx(1090) H323 H323Transport::Close 3:35.882 H225 Caller:81007e8 h323ep.cxx(1681) H323 Clearing connection ip$localhost/17263 reason=EndedByTransportFail 3:35.883 H225 Caller:81007e8 h323.cxx(1634) H225 Signal channel closed. 3:35.897 H323 Cleaner transports.cxx(1172) H323 H323Transport::CleanUpOnTermination for H225 Caller:81007e8 /G __ Do you Yahoo!? Yahoo! Mail is new and improved - Check it out! http://promotions.yahoo.com/new_mail ___ 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 ___ 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] new bounty for modifying calling card application to mysql
I canhelp on this work for free. no problem De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Richard CookEnviado el: Jueves, 05 de Agosto de 2004 03:17 p.m.Para: [EMAIL PROTECTED]Asunto: RE: [Asterisk-Users] new bounty for modifying calling card application to mysql SW, let me know if you find anyone. I'd be willing to contribute some funds to make it work properly. :-) --Richard Cook[EMAIL PROTECTED]Tel: 705-497-9320 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sathya WeerasooriyaSent: Thursday, August 05, 2004 2:04 PMTo: [EMAIL PROTECTED] Digium. ComSubject: [Asterisk-Users] new bounty for modifying calling card application to mysql Hi, I've just initiated a new bounty for the above; http://www.voip-info.org/wiki-Asterisk+bounty+callingcard+to+MySQL Any takers or any contributors please respond to me privately. I do not know exactly how the bounty process works, but I can coordinate on this ? SW
RE: RE: [Asterisk-Users] Best Linux for Asterisk
I think same... All distributions are based on same kernels... And in my opinion, Kernel is who does all work in an operative systemm.. I am wrong?... Actually I am running 3 * boxes in 3 Machines with Redhat 9.0, all are Athlon based. I had some problems, but generally those problem was related to bugs on * and not on Linux.. I have some friends that test Asterisk using Gentoo and Debian, with success results... So just select distro what you feel more comfortable... Regards Sebastian Nocetti -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Walt Reed Enviado el: Jueves, 29 de Julio de 2004 10:12 a.m. Para: [EMAIL PROTECTED] Asunto: Re: RE: [Asterisk-Users] Best Linux for Asterisk On Wed, Jul 28, 2004 at 10:23:41PM +, Mark Woods said: No, it won't be the absolute latest code, but the Debian community is pretty good about keeping packages updated. ah! ah! ah! really... oh oh, so why debian is eons later in releasing new packages... perhaps you're speaking of -unstable debian... that's wy too unstable. A...but I *am* running unstable! And it's been quite, well, stable! :) There is a huge misconception about stable vs unstable. FWIW, I have found debian unstable to be more stable than most other distro's stable releases. For a truely unstable version, experimental would be it. Most of the unstable behavior has been in GUI based parts: Gnome in particular. Since no sane person runs * on a machine that is also running X, it's a non-issue. I've been running * on unstable for about 6 months now with zero downtime other than a few upgrades. Ditto for about a dozen other servers doing high-volume mail, web serving, etc. I find stable unsuitable for most things as all the packages and libraries are too outdated. Yes, the backports help, but then you are not really running stable anymore are you? There are too many dependancies now on other software that needs to be up to date in order to function properly and have the features needed. Anyway, I don't think that it's possible to have a best linux to run any kind of server on. They are all damn good. The core of any linux distro handling non-gui based server applications is virtually identical. Most of the differences are package versions, minor configuration tweaks, package management, and other non-important (when it comes to stability) factors. Do your own research and find one you are comfortable with. For a platform with long-term stability where packages are not constantly changing, maybe something like WhiteBox Linux which is based on the RedHat Enterprise would be appropriate. ___ 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 ___ 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] still can't load oh323 - Are we not supporting H.323 any more?
Read README in oh323 directory, use exactly libraries you can read there, and obviusly apply patch first... Then run ldconfig Put variables on environment And all is ok -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de ruixun wu Enviado el: Viernes, 23 de Julio de 2004 01:09 p.m. Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] still can't load oh323 - Are we not supporting H.323 any more? Hi Michael, I do check the mail list for this problem. But all the answer is using incorrect libraries at run time. And didn't give more details. I just don't know how to use the correct libraries at run time. Now I have installed Pwlib 1.6.6 and Openh323 1.13.5, why do you say I am using incorrect libraries? How to know which libraries are using at run time? --- Michael Manousos [EMAIL PROTECTED] wrote: The problem you experience looks like a usage of incorrect libraries at run time. This problem has been reported and solved many times in the past, through this mailing list but I guess that you didn't check. Check again the libraries (OpenH323/Pwlib) that you use at run time. Michael. Kanuri, Seshu wrote: Why is no one suggesting any solution here for this problem, which has been lingering for a while. Are we not supporting H.323 on Asterisk? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of ruixun wu Sent: Thursday, July 22, 2004 4:06 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] still can't load oh323 hi, I use Redhad 9.0, Pwlib 1.6.6, Openh323 1.13.5, Oh323 0.6.3a. The installation of these package are ok. But when loading oh323, there is error: [chan_oh323.so]Jul 13 12:56:47 WARNING[1074464512]: loader.c:240 ast_load_resource: /usr/local/lib/liboh323wrap.so: undefined symbol: _ZTI14PAbstractArray Jul 13 12:56:47 WARNING[1074464512]: loader.c:421 load_modules: Loading module chan_oh323.so failed! I have asked this question in the mail list before. Someone solved the problem. He said it's due to a problem with redhat 9.0 and need to use the latest glibc and libssl. But the latest version of glibc is 2.3.2, and the glibc in redhat 9 is also 2.3.2. I tried to download the glibc and comply it. But still can't work. So where is the problem? Could you give me more details to help me solv this problem? Thanks a lot Rui __ Post your free ad now! http://personals.yahoo.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 ___ 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 ___ 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 __ Post your free ad now! http://personals.yahoo.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 ___ 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] codec translate
To translate with g729 you need licenses... -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Brent Franks Enviado el: Martes, 20 de Julio de 2004 10:01 a.m. Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] codec translate HI ALL; Is astersik enable to translate between different codecs. I have couple of SIP-UA , one with (a-law) and the other with (g729), registered with my astersik box.Can astersik translate between alaw-g729 and vice varsa. Yes. Also, Google works pretty good too. A simple Google Search for: Asterisk Translate Codec, would have returned a lot of useful searches. I included the link below in case you didn't know where/what google is. http://www.google.com/search?hl=enlr=ie=UTF-8q=Asterisk+Translate+codecb tnG=Search The sixth result looks like a winner. Additionally, Read the WiKi. http://www.voip-info.org - Brent ___ 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 ___ 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
[Asterisk-Users] STILL NO AUDIO
I cant do SIP - CHAN_H323 transmit audio!!! I can hear rings, but when connected, NOTHING It happened in both: SIP - CHAN_H323 and CHAN_H323 - SIP... when it will be solved?
RE: [Asterisk-Users] STILL NO AUDIO
No NAT, no FW, no nothing... from cisco 5300 with public ip without FW, to * with public ip without FW using SIP, and then from * to cisco 5300 without FW using chan_h323 De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de brianEnviado el: Lunes, 19 de Julio de 2004 11:39 a.m.Para: [EMAIL PROTECTED]Asunto: RE: [Asterisk-Users] STILL NO AUDIO Happen to have any NAT in the mix? bkw -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sebastian NocettiSent: Monday, July 19, 2004 9:25 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] STILL NO AUDIO I cant do SIP - CHAN_H323 transmit audio!!! I can hear rings, but when connected, NOTHING It happened in both: SIP - CHAN_H323 and CHAN_H323 - SIP... when it will be solved?
RE: [Asterisk-Users] STILL NO AUDIO
I WANT TO USE G729, I HAVE TO USE IT... -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Eric Wieling Enviado el: Lunes, 19 de Julio de 2004 11:46 a.m. Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] STILL NO AUDIO I suspect it will be solved when you put disallow=all and allow=ulaw in sip.conf and h323.conf (and NO OTHER ALLOW= LINES) On Mon, 2004-07-19 at 09:25, Sebastian Nocetti wrote: I cant do SIP - CHAN_H323 transmit audio!!! I can hear rings, but when connected, NOTHING It happened in both: SIP - CHAN_H323 and CHAN_H323 - SIP... when it will be solved? -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ 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 ___ 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] STILL NO AUDIO
Testing both... -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Michael Manousos Enviado el: Lunes, 19 de Julio de 2004 12:25 p.m. Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] STILL NO AUDIO Why don't you use asterisk-oh323? Michael. Sebastian Nocetti wrote: I WANT TO USE G729, I HAVE TO USE IT... -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Eric Wieling Enviado el: Lunes, 19 de Julio de 2004 11:46 a.m. Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] STILL NO AUDIO I suspect it will be solved when you put disallow=all and allow=ulaw in sip.conf and h323.conf (and NO OTHER ALLOW= LINES) On Mon, 2004-07-19 at 09:25, Sebastian Nocetti wrote: I cant do SIP - CHAN_H323 transmit audio!!! I can hear rings, but when connected, NOTHING It happened in both: SIP - CHAN_H323 and CHAN_H323 - SIP... when it will be solved? ___ 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 ___ 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] STILL NO AUDIO
What kind of problem? All works OK except that config -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Holger Schurig Enviado el: Lunes, 19 de Julio de 2004 12:32 p.m. Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] STILL NO AUDIO I WANT TO USE G729, I HAVE TO USE IT... When you have no FW and no NAT, then you seem to be inside your local network. In this case you shouldn't really care ?!?! ___ 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 ___ 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
[Asterisk-Users] Astersik with g729 and 120 active channels with digium card ISDN PRI
Hello, I want to know what kind of equipment I need to handle 120 simultaneous calls with a Digium 4E1 card... and using 120 G.729 licences some help? thanks Sebastian.
RE: [Asterisk-Users] How to uninstall Asterisk?
IN MY HONEST OPINION... IMHO I am right? -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de ruixun wu Enviado el: Miércoles, 14 de Julio de 2004 11:07 a.m. Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] How to uninstall Asterisk? hi Gus and Roger, Thanks for you reply. I choose no load the chan_oh323. The asterisk now can start again. :) And Gus, could you tell me what's the meaning of IMHO? I can't find the topic about IMHO in WIFI. Thanks a lot! Best Regards Rui --- CW_ASN [EMAIL PROTECTED] wrote: Hi, After I install openh323, the asterisk cann't work anymore. Asterisk failed in loading chan_oh323. I cann't deleted the openh323 package, so the only thing I can do is to reinstall Asterisk. I checked out the asterisk and make install Astersik without installed openh323, but when I started Asterisk, Asterisk still loaded openh323 and failed. Does anyone know how to uninstall Asterisk? If you don't like to load a channel or module, you can choose for two methods: - You can delete it. The channels and apps are located in /usr/lib/asterisk/modules. - You can choose to not load when asterisk loads. Use modules.conf, set noload = foo.so At least is strage... I'm using chan_oh323 without failures, and IMHO, it's more stable and powerful than others. I'm not wish to start a war, it's just my opinion. Regards, Gus ___ 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 __ Post your free ad now! http://personals.yahoo.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 ___ 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
[Asterisk-Users] CHAN_H323 bridge SIP no audio
I tried a lot of times to get it worked, but I cant obtain audio using SIP-chan_h323 or chan_h323-SIP I tried disbling FastStart without good results... What's the problem? I need to do BRIDGE between SIP and H.323!! help!! Sebastian.-
RE: [Asterisk-Users] How to uninstall Asterisk?
TODO OK!!!... Che, de donde te ubico? -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Gonzalo Servat Enviado el: Miércoles, 14 de Julio de 2004 01:15 p.m. Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] How to uninstall Asterisk? Correcto, I think it's also In My Humble Opinion too. Gonzalo P/D: Como andas Seba... :) On Wed, 2004-07-14 at 11:45 -0300, Sebastian Nocetti wrote: IN MY HONEST OPINION... IMHO I am right? -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de ruixun wu Enviado el: Miércoles, 14 de Julio de 2004 11:07 a.m. Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] How to uninstall Asterisk? hi Gus and Roger, Thanks for you reply. I choose no load the chan_oh323. The asterisk now can start again. :) And Gus, could you tell me what's the meaning of IMHO? I can't find the topic about IMHO in WIFI. Thanks a lot! Best Regards Rui --- CW_ASN [EMAIL PROTECTED] wrote: Hi, After I install openh323, the asterisk cann't work anymore. Asterisk failed in loading chan_oh323. I cann't deleted the openh323 package, so the only thing I can do is to reinstall Asterisk. I checked out the asterisk and make install Astersik without installed openh323, but when I started Asterisk, Asterisk still loaded openh323 and failed. Does anyone know how to uninstall Asterisk? If you don't like to load a channel or module, you can choose for two methods: - You can delete it. The channels and apps are located in /usr/lib/asterisk/modules. - You can choose to not load when asterisk loads. Use modules.conf, set noload = foo.so At least is strage... I'm using chan_oh323 without failures, and IMHO, it's more stable and powerful than others. I'm not wish to start a war, it's just my opinion. Regards, Gus ___ 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 __ Post your free ad now! http://personals.yahoo.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 ___ 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 ___ 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 ___ 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] chan_oh323
ldconfig, check that /etc/ld.so.conf have path to where oh323 library is and then run ldconfig De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Fathallah SoumayaEnviado el: Martes, 13 de Julio de 2004 12:27 p.m.Para: [EMAIL PROTECTED]Asunto: Re: [Asterisk-Users] chan_oh323 Hello, I have been trying for a while to make the oh323 channel working but i didnt manage, i have everything compiled correctly but asterisk find somethign like an "undefined symbol" when it loads the oh323 module... i dont know if u have seen this before, I am deseperate to find the solution , i am involved in a very important project and i am out of time :( I would be very grateful if you can help me... Best Regards, soumayaLars Degenhardt [EMAIL PROTECTED] wrote: Hello,has anybody managed to register with two gatekeepers usingchan_oh323?Lars___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Créez gratuitement votre Yahoo! Mail avec 100 Mo de stockage ! Créez votre Yahoo! Mail Dialoguez en direct avec vos amis grâce à Yahoo! Messenger !
[Asterisk-Users] NO AUDIO IN BOTH DIRECTIONS
hello all, I am having a trouble with Audio using h.323 channel... I am doing this Call comes into cisco 5300 and is sent to Asterisk, asterisk catch call with h.323 driver and send call to a SoftSwitch that routes the call, I can see log debug telling me, CALLED XXX, and then RINGING, and I can hear ring tones... but when call is answered, I DONT HEAR ANYTHING... I am using lastest ASTERISK download somebody can help me to solve this problem thanks..!!
[Asterisk-Users] G.723.1
Title: Mensaje Hi, I want to use G.723.1 on *, I read it is supported in Pass Through mode, but I don't understand whats the meaning of that. I have a GW 5300 and an ATA 186 and I want to place calls to PSTN. I setup this config: [general]port = 5060 bindaddr = xx.xx.xx.xx context = sip tos=throughput maxexpirey=360 defaultexpirey=120 [gw5300]type=friendinsecure=yeshost=xx.xx.xx.xxdisallow=allallow=g723allow=ulawcanreinvite=noreinvite=nodtmfmode=rfc2833 [1500]type=friendusername=1500secret=x disallow=allallow=g723allow=ulawhost=dynamiccanreinvite=noqualify=300dtmfmode=rfc2833 and this extension.conf [sip] exten = _0114XXX,1,Dial(SIP/[EMAIL PROTECTED]:5060) where xx.xx.xx.xx is GW ip address but when I place a call from ATA to GW, telephone rings and inmediatly hangs when person answer the phone. When I use only ULAW, all works OK. somebody can tell what I am missing?. someone can help configuring * to use G723 pass through
[Asterisk-Users] Radius on *
Does Asterisk support Radius accounting? -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de [EMAIL PROTECTED] Enviado el: Lunes, 17 de Noviembre de 2003 12:08 p.m. Para: [EMAIL PROTECTED] Asunto: Asterisk-Users digest, Vol 1 #1912 - 11 msgs Send Asterisk-Users mailing list submissions to [EMAIL PROTECTED] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. RE: FXO Cards in Australia (Gonzalo Servat) 2. Re: IAX2 connectivity problem (qualify=yes) (Philipp von Klitzing) 3. RE: FXO Cards in Australia (Bryan Nolen) 4. RE: FXO Cards in Australia (Gonzalo Servat) 5. Re: Meetme : Zaptel ztdummy errors (Andrew Thompson) 6. SIP soft phone registration (Steve Murphy) 7. Re: DTMF (Sean P. Robertson) 8. VOIP phonesets vs. cheap Analog touch-tone sets with Asterisk (Steve Murphy) 9. Re: NuFone International Calls (marrandy) 10. Re: Meetme : Zaptel ztdummy errors ([EMAIL PROTECTED]) 11. iconnecthere incoming ([EMAIL PROTECTED]) --__--__-- Message: 1 Subject: RE: [Asterisk-Users] FXO Cards in Australia From: Gonzalo Servat [EMAIL PROTECTED] To: [EMAIL PROTECTED] Organization: Webtastic Date: Tue, 18 Nov 2003 00:17:30 +1100 Reply-To: [EMAIL PROTECTED] On Mon, 2003-11-17 at 23:53, Adam Goryachev wrote: Yes, echo problems do still exist, I would suggest testing it before going live. Yeah, so I've heard. A couple of points to note: 1) Using soft phones seems to compound the issue So the echo problems are not so bad when using software phones? 2) A faster CPU seems to help (I upgraded from a PII300 to a Athlon 2200) 3) When dialling in/out over the ISDN DTMF won't work (at least I haven't seen the patch which purportedly allows it to work) when you use the isdn4linux patch. This is specific to the NetJet card once again, right? Time to go hunting for the patch... 4) Without the above kernel patch you will hear DTMF tones instead of the other persons voice when they talk. They don't hear the tones or notice anything wrong. Hmm, not good. Since we want to run a small IVR the DTMF tones are kinda needed. In short, if you can live with the above problems, then you can get away with it, from what I know now, I would suggest getting a chan_capi capable device, though I haven't tried that yet. The NetJet is supposedly CAPI capable. Have you tried installing this? -- http://www.junghanns.net/asterisk/page1.html I am about to switch from a netjet card to a TE4xxP card as soon as possible, I have a OnRamp 10 being installed tomorrow. This is largely to increase the number of incoming lines, but partly to resolve the above issues, and also partly to try to resolve long running reliability issues which may in fact be related to the TDM400P anyway. In which case I will be looking for a T1 channel bank some time soon :( Argh, the fun never stops :) PS, I have a brand new Traverse Netjet card available (it was to be used for a dial-up ISDN internet account) which is no longer needed. How much do you want for it? If you can confirm whether the capi channel driver works with it and reduces the echo problem, I'll be interested. Thanks for your help. Regards, Gonzalo --__--__-- Message: 2 Date: Mon, 17 Nov 2003 14:28:14 +0100 From: Philipp von Klitzing [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] IAX2 connectivity problem (qualify=yes) To: [EMAIL PROTECTED] Organization: AEGEE Reply-To: [EMAIL PROTECTED] Hi! Try qualify=3000 which will increase the time between checks.. Although it sounds like there is more to this problem than just increasing the time.. That's not really what I want to do - quality is really bad if you go above 2000, so it makes sense to keep it at this. I can schedule a reload, of course, e.g. once an hour, but that can't be the correct way to do this... is there really no-one else out here who has seen this?? Cheers, Philipp --__--__-- Message: 3 From: Bryan Nolen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FXO Cards in Australia Date: Tue, 18 Nov 2003 00:48:34 +1100 Reply-To: [EMAIL PROTECTED] Re: these problems with the NetJet Cards: have people spoken with Traverse about them? I have found them to be most helpful with any problems (mainly with the Pulsar PCI ADSL cards) Try talking to [EMAIL PROTECTED] ? -Bryan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gonzalo Servat Sent: Tuesday, 18 November 2003 12:18 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FXO Cards in Australia On Mon, 2003-11-17 at 23:53, Adam
[Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #1918 - 9 msgs
An example for Radius is calling cards.. I can use * for this kind of service... With platforms that use Radius Server. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de [EMAIL PROTECTED] Enviado el: Lunes, 17 de Noviembre de 2003 07:16 p.m. Para: [EMAIL PROTECTED] Asunto: Asterisk-Users digest, Vol 1 #1918 - 9 msgs Send Asterisk-Users mailing list submissions to [EMAIL PROTECTED] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. RE: wireless ([EMAIL PROTECTED]) 2. Re: Radius on * (Jeremy McNamara) 3. Re: IAX2 connectivity problem (qualify=yes) (Andrew Thompson) 4. Re: VOIP phonesets vs. cheap Analog touch-tone sets with Asterisk (Howard White) 5. Re: SIP calls no longer work (Andrew Thompson) 6. 3Com NBX phones (Andrew Nelson) 7. asterisk and Codec G-723 (Javier Rios) 8. Re: DTMF (Scott England) --__--__-- Message: 1 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] wireless Date: Tue, 18 Nov 2003 08:03:06 +1030 Reply-To: [EMAIL PROTECTED] Don't sound bad do they Except their ship date is mid January 2004 Regards Mick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Iain Stevenson Sent: Monday, 17 November 2003 10:34 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] wireless AFAIK the 7920 needs CallManager to work - if you haven't got that you'll have to wait for Cisco to make a general purpose version - or maybe buy a Pulver phone http://www.pulverinnovations.com/ - assuming that works with * Iain --On Monday, November 17, 2003 6:31 am -0500 Jeremy McNamara [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: Has anyone got a mobile wireless phone working with * yet Is it possible to use the Cisco 7920 with skinny Not sure, send me one and I'll test it for you. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --__--__-- Message: 2 Date: Mon, 17 Nov 2003 16:33:10 -0500 From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Radius on * Reply-To: [EMAIL PROTECTED] Sebastian Nocetti wrote: Does Asterisk support Radius accounting? No and there is absolutely no need for it to. RADIUS is not anything that should have ever been deployed in a VoIP environment. There are many methods to talk directly to a database, why add another layer of complexity and point of failure? Jeremy McNamara --__--__-- Message: 3 From: Andrew Thompson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] IAX2 connectivity problem (qualify=yes) Date: Mon, 17 Nov 2003 16:54:06 -0500 Reply-To: [EMAIL PROTECTED] LS0tLS0gT3JpZ2luYWwgTWVzc2FnZSAtLS0tLSANCkZyb206ICJPbGxlIEUuIEpvaGFuc3Nv biIg PG9lakBlZHZpbmEubmV0Pg0KVG86IDxhc3Rlcmlzay11c2Vyc0BsaXN0cy5kaWdpdW0uY29t Pg0K U2VudDogTW9uZGF5LCBOb3ZlbWJlciAxNywgMjAwMyAzOjU4IFBNDQpTdWJqZWN0OiBSZTog W0Fz dGVyaXNrLVVzZXJzXSBJQVgyIGNvbm5lY3Rpdml0eSBwcm9ibGVtIChxdWFsaWZ5PXllcykN Cg0K DQo+IFdpcGVPdXQgd3JvdGU6DQo+IA0KPiA+IFBoaWxpcHAgdm9uIEtsaXR6aW5nIHdyb3Rl Og0K PiA+IA0KPiA+PiBIaSB0aGVyZSwNCj4gPj4NCj4gPj4gSSBzdGlsbCBoYXZlIGlzc3VlcyB3 aXRo IHRoZSBJQVggY29ubmVjdGlvbiBiZXR3ZWVuIHR3byBzZXJ2ZXJzIChvbmUgDQo+ID4+IHN0 YXRp YyAoc2VydmVyIEEpLCBvbmUgZHluYW1pYyAoc2VydmVyIEIpLCBub25lIGJlaGluZCBOQVQp Og0K PiA+Pg0KPiA+PiBCIHJlZ2lzdGVycyB3aXRoIEEsIGFuZCAiaWF4MiBzaG93IHJlZ2lzdHJ5 IiBz aG93cyB0aGF0IGV2ZXJ5dGhpbmcgaXMgDQo+ID4+IGZpbmUuIEhvd2V2ZXIsIGFmdGVyIGEg d2hp bGUgaWYgSSBjaGVjayBvbiBzZXJ2ZXIgQSB3aXRoICJpYXgyIHNob3cgDQo+ID4+IHBlZXJz IiBJ IHNlZSBhIHN0YXR1cyBvZiBVS05PV04gKGluIGlheC5jb25mIHRoZXJlIGlzIGEgcXVhbGlm eT15 ZXMgDQo+ID4+IHN0YXRlbWVudCBmb3Igc2VydmVyIEIpLg0KPiA+Pg0KPiA+PiBUaGUgcHJv Ymxl bSBpcyB0aGF0IHRoZSBjb25uZWN0aW9uIHRvIHNlcnZlciBCIHJlZ3VsYXJseSBnZXRzIHdv cnNl IA0KPiA+PiB0aGFuIDIwMDAgbXMgYW5kIGlzIHRodXMgdGVtcG9yYXJpbHkgbGlzdGVkIGFz IHVu YXZhaWxhYmxlIC0gZmluZSBzbyANCj4gPj4gZmFyLCB0aGF0IGlzIGNvcnJlY3QuIEJ1dCBh dCBz b21lIHBvaW50IHNlcnZlciBBIHN0b3BzIHRvIHBpY2sgdXAgdGhlIA0KPiA+PiB3b3JraW5n ICgh KSBjb25uZWN0aW9uIGFuZCBwZXJtYW5lbnRseSBzaG93cyBzdGF0dXMgVU5LT1dOIGFsdGhv dWdo IA0KPiA+PiB0aGUgY29ubmVjdGlvbiBpcyBmaW5lIGFnYWluICg0MCBtcyBvciBzbykuIEl0 IGRv ZXMgaGVscCB0byBpc3N1ZSBhIA0KPiA+PiAicmVsb2FkIiBvbiBzZXJ2ZXIgQSwgYnV0IHRo YXQn cyBmb3Igc3VyZSBub3QgdGhlIHdheSBpdCBzaG91bGQgYmUuLi4gDQo+ID4+IHdpdGhvdXQg dGhh
[Asterisk-Users] Media Negotiation Failed
Title: Mensaje Hi, I have this scenario Cisco 5300 (public ip. 200.47.xx.xx) --- Asterisk (public ip: 64.76.xx.xx) -- Cisco 3600 (public ip: 64.76.xx.xx , same network than * ) When a calls comes in Cisco 5300, this send this calls with SIP to *, asterisk plays a welcome message and resend call to Cisco 3600 that have 4 analog lines connected... but after cisco play welcome message and whensend SIP to 3600, I have this error: v=0o=root 20045 20045 IN IP4 64.76.xx.xx - asterisk ip addresss=sessionc=IN IP4 64.76.xx.xx - asterisk ip address.t=0 0m=audio 15372 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16(no NAT) to 64.76.xx.xx:5060 - 3600 ip addressSip read: LISIP/2.0 400 Bad Request - 'Media Negotiation Failed'Via: SIP/2.0/UDP 64.76.xx.xx:5060;branch=z9hG4bK31ba01da - asterisk ip addressFrom: "1143724956" sip:[EMAIL PROTECTED];tag=as33c45436 - * ip addressTo: sip:[EMAIL PROTECTED] -3600 ip addressCall-ID: [EMAIL PROTECTED]Warning: 304 64.76.xx.xx:0 "Media Type(s) Unavailable" - 3600 ip addressCSeq: 102 INVITE then I have too another GW 5300, with same IOS and same config.. and with it, all work OK!!!... I don't understand what is the problem!!... IT WORKS OK!!!.. Cisco 5300 (public ip. 64.76.xx.xx) --- Asterisk (public ip: 64.76.xx.xx) -- Cisco 3600 (public ip: 64.76.xx.xx , same network than * ) Some clue?
[Asterisk-Users] RE: Media Negotiation Failed
Codecs are g711ulaw, on both Cisco5300... Dial Peer config is showed below Los codecs que uso son G711ulaq, en los dos Cisco5300, te muestro los dialpeers... GW that not work - GW que no funciona translation-rule 1017 Rule 0 8002666333 1000 dial-peer voice 1016 voip destination-pattern 8002666333 translate-outgoing called 1017 session protocol sipv2 session target ipv4:64.76.xx.xx --- IP DE ASTERISK. dtmf-relay h245-alphanumeric codec g711ulaw no vad GW that work - GW que funciona translation-rule 7 Rule 0 ^3104 1000 Rule 1 ^3105 1000 dial-peer voice 7 voip destination-pattern 310[4-5] translate-outgoing called 7 session protocol sipv2 session target ipv4:64.76.xx.xx IP DE ASTERISK. dtmf-relay h245-alphanumeric codec g711ulaw no vad -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de [EMAIL PROTECTED] Enviado el: Miércoles, 12 de Noviembre de 2003 02:12 p.m. Para: [EMAIL PROTECTED] Asunto: Asterisk-Users digest, Vol 1 #1869 - 11 msgs Send Asterisk-Users mailing list submissions to [EMAIL PROTECTED] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. Re: DIAX 0.93 with some sound improvements and not only... (Ariel Batista) 2. Re: OT : For the SQL gurus.. (Roy Sigurd Karlsbakk) 3. Re: Media Negotiation Failed (CW_ASN - Gus) 4. Re: DIAX 0.93 with some sound improvements and not only... (Dan) 5. Re: OT : For the SQL gurus.. (Tilghman Lesher) 6. Re: DIAX 0.93 with some sound improvements and not only... (reseaux) 7. Re: OT : For the SQL gurus.. (WipeOut) 8. Re: OT : For the SQL gurus.. (WipeOut) 9. TAPI development (Michael Devenijn) 10. Re: OT : For the SQL gurus.. (Ernest W. Lessenger) 11. Dial Plan Sequencing (Stephen R. Besch) --__--__-- Message: 1 Date: Wed, 12 Nov 2003 10:50:05 -0500 From: Ariel Batista [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] DIAX 0.93 with some sound improvements and not only... Reply-To: [EMAIL PROTECTED] -- Original Message -- From: Dan [EMAIL PROTECTED] Hi all, DIAX 0.9.3 is available for download from the same place: http://www.laser.com/dante or http://www.geocities.com/tdanro Thank you for the update! I have the following problems with it! When exiting the program we get a General Protech error. Also when calling Zap ports it keeps ringing. From DIAX to Sip it works fine! It actually sound better then before! But I can not call it from SIP get Audio missmatch. I can call it from normal Zap ports! Hope this helps! Keep up the work! --__--__-- Message: 2 Date: Wed, 12 Nov 2003 17:01:10 +0100 (CET) From: Roy Sigurd Karlsbakk [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] OT : For the SQL gurus.. Reply-To: [EMAIL PROTECTED] Thanks everyone for your help on this.. For those who are interested I have done some speed tests on these two queries (below) on my server and the results are.. Test script of 1000 quieries.. Query1 (code field not indexed) = 47.183s Query1 (code field indexed) = 45.731s Query2 (code field not indexed) = 109.321s Query2 (code field indexed) = 2.302s Tried fulltext indexing? --__--__-- Message: 3 From: CW_ASN - Gus [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Media Negotiation Failed Date: Wed, 12 Nov 2003 13:01:29 -0300 Reply-To: [EMAIL PROTECTED] This is a multi-part message in MIME format. --=_NextPart_000_0061_01C3A91D.16D9E3F0 Content-Type: text/plain; charset=iso-8859-1 Content-Transfer-Encoding: quoted-printable MensajeFijate en los 'voice codecs' de los dial-peers. - Original Message -=20 From: Sebastian Nocetti=20 To: [EMAIL PROTECTED] Sent: Wednesday, November 12, 2003 12:41 PM Subject: [Asterisk-Users] Media Negotiation Failed Hi, I have this scenario Cisco 5300 (public ip. 200.47.xx.xx) --- Asterisk (public ip: = 64.76.xx.xx) -- Cisco 3600 (public ip: 64.76.xx.xx , same network than = * ) When a calls comes in Cisco 5300, this send this calls with SIP to *, = asterisk plays a welcome message and resend call to Cisco 3600 that have = 4 analog lines connected... but after cisco play welcome message and = when send SIP to 3600, I have this error: v=3D0 o=3Droot 20045 20045 IN IP4 64.76.xx.xx - asterisk ip address s=3Dsession c=3DIN IP4 64.76.xx.xx - asterisk ip address. t=3D0 0 m=3Daudio 15372 RTP/AVP 0 101 a=3Drtpmap:0 PCMU/8000 a=3Drtpmap:101 telephone-event/8000 a=3Dfmtp:101 0-16 (no NAT) to 64.76.xx.xx:5060 - 3600 ip address Sip read: LI SIP/2.0 400 Bad Request
