[Asterisk-Users] Rate Engine Examples
Anyone has an example of how a working record for agress and rates tables should look? I'been trying all the thinkable patterns, obviously not the right ones, for the last two days. Tkx, LTenorio ___ 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] PIX!!!!!
Chris, I suggest the same, but in case you want to use the fixup feature http://www.cisco.com/en/US/products/hw/vpndevc/ps2030/products_configura tion_example09186a00801fc74a.shtml LTenorio -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julio Arruda Sent: Thursday, January 20, 2005 7:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] PIX! Christopher wrote: Can anyone point me in a good direction for configuring SIP through a PIX using 1:1 NAT. I have read anything I could get my hands on and tried them all with very little success. I can get it to work through the cheap little cable modem routers, but not this PIX. I -can- make a direct SIP call using the IP address of the * server ([EMAIL PROTECTED]), but when I do that * still doesn't show it registering. Even when I call through this method the phone comes up as UNREACHABLE and the port is listed as 0 instead of 5060 like all the internal phones. I seem to recall some weird thing in the PIX, where you had to disable the SIP fixup to work (and of course, to use some nat traversal trick, like outbound proxy). ___ 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] RE: [Asterisk-biz] Guatemala DID's?
In the next couple of weeks we will be starting the beta phase of our Guatemala POP. If you could wait, welcome. LTenorio -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Phil Astin Sent: Sunday, January 16, 2005 6:23 PM To: [EMAIL PROTECTED]; asterisk-users@lists.digium.com Subject: [Asterisk-biz] Guatemala DID's? I'm looking for a company that offers Guatemala DID's. I saw that Lingo does, but Lingo isn't easily compatible w/ Asterisk, so they're a last resort. Thanks in advanced, Phil Astin. ___ Asterisk-Biz mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-biz ___ 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: FW: [Asterisk-Users] Radius on *
Any other source?, I'm getting not found on that host -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Tkachuk Sent: Monday, January 17, 2005 6:06 PM To: asterisk-users@lists.digium.com Subject: Re: FW: [Asterisk-Users] Radius on * Hello all, It's my try to make some 'emulation' of vovida's b2bua using asterisk. I was in rush while writing it, so I sure there is much code that can be cleaned, great that not too much. :) http://dslmax.boom.ru/asterisk_b2bua_v0.1.zip cdrradius and agi script inside. __ Mike Tkachuk ___ 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] Radius on *
I'm currently trying to use a Radius server for acct and auth, cause much of our systems are using it. Anyone has an asterisk server working with Radius Auth and Acct? Tkx, LTenorio ___ 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] PRI concentrator
IMHO, In your environment I would recommend to use several Gateways (Cisco, Quintum, whatever you choose) to concentrate and do the transcoding, and use * to switch them and do the other stuff you need. A lot of times where seen in the list people that measure the load on * servers that cannot do more than 60 channels at a time. And personally a do trust more on a GW than in * to do PRI. LTenorio - Original Message - From: Michael B. Murdock [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, January 13, 2005 3:27 PM Subject: Re: [Asterisk-Users] PRI concentrator We are looking to solve the same problem for different reasons.. I am currently looking into the Lucent (Ascend) MAX TNT to terminate the PRI's and then use VoIP/SIP to connect to the Asterisk platform. BTW: Who makes your class-3 switch you are having so much trouble with? -- Mike - Original Message - From: Matthew Boehm [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, January 13, 2005 2:24 PM Subject: [Asterisk-Users] PRI concentrator Hey gang, We currently have a class 3 switch (CSX) that..well..it sucks. It does terrible CDR writes, doesn't support LCR, the list goes on and on. We want to replace this with several asterisk boxes each running one or two 4 port PRI cards. The problem is: I can plug in 20 PRI lines into the CSX (from PSTN) and have 1 come from CSX into asterisk. If 1 call comes in on each of the 20 pris, the CSX concentrates them into 1 pri into asterisk. Thus I only need a single PRI card for asterisk box. If I want to completly replace the CSX, I would need several boxes with 1 or 2 4 port cards. But that is a big waste of money when I can concentrate into less cables. Does anyone know of some piece of hardware (other than a carrier switch) that will do the same thing? Keeping our current switch doesn't cost anything except in terms of power consumption and physical space (its like 5U or 8U). Thanks, Matthew ___ 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 ___ 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] Cant receive calls after network goes down and up
That's probably a timeout problem in the nat
box.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
NortonSent: Wednesday, January 12, 2005 6:44 PMTo:
'Asterisk Users Mailing List - Non-Commercial Discussion'Subject:
[Asterisk-Users] Cant receive calls after network goes down and
up
Hi,
I have several Grandstream phones connected to Asterisk, some behind NAT
and others not. If I reboot all the phones, everything is fine. Should the
connection go down, and then come back again, those behind a NAT are still able
to make calls, but are unable to receive calls.
-- Executing Dial("SIP/1239-ba74", "SIP/1242|60|t") in new
stack
Jan 12 23:45:19 NOTICE[21576]: app_dial.c:803 dial_exec: Unable to create channel of type 'SIP' (cause
3)
== Everyone is
busy/congested at this time (1:0/1/0)
However, extension 1242 is still
able to call 1239?
Is this a configuration problem in
Asterisk or in the phones?
Please
help
Regards
David
Norton
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RE: [Asterisk-Users] chan_oh323 Module for Asterisk
I got it, but email it to the list is not a good option. Who 're interested just email me, I'll send it asap. But AFAIK, you still need the wrapper. LTenorio -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT) Sent: Wednesday, January 05, 2005 5:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] chan_oh323 Module for Asterisk If anyone in the list has a working version of the chan_oh323.so file for Fedora Core 2 and Redhat, can he email the same to the list as attachment. This will reduce the pain for many of the users who are trying to compile the same from the libraries, which never seemed to work. Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Humberto Aicardi Sent: Wednesday, January 05, 2005 3:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] chan_oh323 gatekeeper Hi folks, Until now I have used only SIP IAX2 with success and understand them pretty well. The point is that someone has asked me to configure an * box for them, the problem is that they want to use H.323. I have already compiled and tested the chan_oh323 with asterisk and works. The problem is that the tests need a gatekeeper, my question is: Do I need always need a gatekeeper? Or my FXO H.323 gateway can register with * ? Thanks, Humberto Aicardi NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ 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] Asterisk-AS5350 misplaced RTP to 127.0.0.1(AS5350 party don't hear)
Try sending 5350 config and oh323.conf, versions, etc...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Goran Dj.
Sent: Wednesday, December 22, 2004 4:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk-AS5350 misplaced RTP to
127.0.0.1(AS5350 party don't hear)
My configuration is:
[ISDNPRI] -- [CISCO AccessServer AS5350] --H.323-- [ASTERISK] --
[CISCO ip phone 12SP+/Skinny]
When call is initiated from IP phone - Asterisk - AS5350 - ISDN
everything working ok (RTP is ok).
But, when call coming from ISDN - AS5350 - Asterisk - IP phone IP
phone party can hear ISDN party, but ISDN (incoming) party canNOT hear
IP phone party because RTP stream from Asterisk is sent to 127.0.0.1
instead to real IP address of AS5350
Here is H.323 debug, for both situations:
1) ---
--- outgoing call (RTP is ok, both party can hear) --
-- Call token is ip$localhost/12862
-- Call reference is 12862
-- Sending SETUP message
Recieved Open Recieve Channel Ack
=*= In CreateRealTimeLogicalChannel for call 12862
-- externalIpAddress: 10.0.3.15
-- externalPort: 14152
-- SessionID: 1
-- Direction: IsReceiver
-- Started logical channel: receiving G.711-uLaw-64k{sw}
-- channelsOpen = 1
=-= In OnAlerting for call 12862: sessionId=1
--- found logical channel. Connecting RTP RTP channel id 1 parameters:
-- remoteIpAddress: 10.10.10.61
-- remotePort: 16862
-- ExternalIpAddress: 10.0.3.15
-- ExternalPort: 14152
-- Ringing phone for 10.10.10.61
-- Asked to indicate 'Remote end is ringing' condition on channel
Skinny/[EMAIL PROTECTED]
RFC3389: 1 bytes, level 4...
Dec 22 18:51:26 NOTICE[557082]: rtp.c:289 process_rfc3389: RFC3389
support incomplete. Turn off on client if possible =*= In
CreateRealTimeLogicalChannel for call 12862
-- externalIpAddress: 10.0.3.15
-- externalPort: 14152
-- SessionID: 1
-- Direction: IsTransmitter
-- Started logical channel: sending G.711-uLaw-64k{sw}
-- channelsOpen = 2
=-= In OnConnectionEstablished for call 12862
-- Connection Established with 10.10.10.61
-- Asked to indicate 'Stop tone' condition on channel
Skinny/[EMAIL PROTECTED]
=-= In OnReceivedAckPDU for call 12862
channelsOpen = 1
2) ---
---incoming call (RTP misplaced, incoming party don't hear)
Sending alerting
-- Received Facility message...
-- Received Facility message...
-- Received Facility message...
-- Received Facility message...
-- Received Facility message...
=*= In CreateRealTimeLogicalChannel for call 5006
-- externalIpAddress: 10.0.3.15
-- externalPort: 17166
-- SessionID: 1
-- Direction: IsReceiver
-- Started logical channel: receiving G.711-uLaw-64k{sw}
-- channelsOpen = 1
RTP channel id 1 parameters:
-- remoteIpAddress: 10.10.10.61
-- remotePort: 16700
-- ExternalIpAddress: 10.0.3.15
-- ExternalPort: 17166
Recieved Open Recieve Channel Ack
answering call
=*= In CreateRealTimeLogicalChannel for call 5006
-- externalIpAddress: 10.0.3.15
-- externalPort: 17166
-- SessionID: 1
-- Direction: IsTransmitter
-- Started logical channel: sending G.711-uLaw-64k{sw}
-- channelsOpen = 2
RTP channel id 1 parameters:
-- remoteIpAddress: 127.0.0.1
-- remotePort: 2070
-- ExternalIpAddress: 10.0.3.15
-- ExternalPort: 17166
=-= In OnConnectionEstablished for call 5006
-- Connection Established with 10.10.10.61
-- Received Facility message...
=-= In OnReceivedAckPDU for call 5006
-- Received Facility message...
channelsOpen = 1
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RE: [Asterisk-Users] Cisco AS5XXX to asterisk debugging.
Pls, post your Cisco and * config files. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jorge Verastegui G Sent: Friday, December 10, 2004 12:30 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco AS5XXX to asterisk debugging. Hi, I have a serious problem to configure Cisco AS5XXX and Asterisk , I trying to use asterisk for PSTN(A) Cisco AS5xxx ASteriskPSTN(B) (No Nat, no Firewall) I hear (on the PSTN(A)) clearly what the other person is saying, but the other person (on the PSTN(B) side) hears nothing from PSTN(A). I use tcpdump for debug de rtp trafic, and ouput contains 19:06:00.741293 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:00.763133 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:00.740415 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:00.810312 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:00.860314 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:00.980351 IP (tos 0x0, ttl 64, id 180, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.000313 IP (tos 0x0, ttl 64, id 181, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.014822 IP (tos 0x68, ttl 255, id 1, offset 0, flags [none], proto 17, length: 164) y.y.y.y.18975 x.x.x.x.19927: UDP, length 136 19:06:01.020312 IP (tos 0x0, ttl 64, id 182, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.040302 IP (tos 0x0, ttl 64, id 183, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.060343 IP (tos 0x0, ttl 64, id 184, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.083311 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.128314 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.130316 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.165318 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.186312 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 Where x.x.x.x = ip address of Astersik y.y.y.y = ip address of Cisco Two types of codecs were proven ( ulow, g729 ). When use the Asterisk with Sip phones everything works well. SipPhone--Asterisk---PSTN(B) The configurations, are the usual ones (from the wiki). the version of asterisk is 1.0.3, the linux is FC2. Please help me. -- Jorge Verastegui G [EMAIL PROTECTED] RedCetus S.R.L. ___ 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] 7905G Firmware
Nope, each phone has it's own firmware. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Marlowe Sent: Friday, December 03, 2004 7:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] 7905G Firmware Is the 7905G Firmware the same as the 7960 firmware? -- MBM ___ 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] Micronet problem
Some more info would be nice What software version? Did you setup the Micronet as peer, proxy or as gateway? Did you setup the user and password? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Federico Gonzalez Sent: Wednesday, December 01, 2004 7:03 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Micronet problem Hello, I connected a Micronet SP5014 2FXS + 2FXO gateway to the asterisk, the problem is i can make call but can't receive calls. If i make a sip show peers it shows the micronet is not connected to the asterisk. Does anybody knows how to configure the micronet and asterisk to solve this problem ? Thank you ___ 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] cisco dial-peer voip
What software version do u've, just 12.3T, support IP2IP feature. I suggest you to use * instead -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, November 30, 2004 10:53 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] cisco dial-peer voip You have 3 dial-peers (40,50,60) all with the same destination-pattern .+ (that means all calls) Think it first tries dial-peer 40 because it has preference 0... And then peers 50 (or) 60 (both preference 5) ... It uses the second preference because the peer 40 just doesn't work And that sounds logically because you have session transport tcp ... And asterisk doesn't support that... Use session transport udp Regards, Niels -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jan Baggen Sent: Tuesday, November 30, 2004 2:36 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] cisco dial-peer voip 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 ___ 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] Where to buy POLYCOM phones?
Has anyone bought from cheapentstuff?¿ LTenorio -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Marlowe Sent: Monday, November 01, 2004 10:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Where to buy POLYCOM phones? http://www.cheapnetstuff.com/ProductDetails.aspx?Prod_ID=190110Source=Froogle Has them for 116.99 On Mon, 18 Oct 2004 20:02:12 -0500, Kristian Kielhofner [EMAIL PROTECTED] wrote: Deon Rodden wrote: Either way. I've bought several devices from b2tech on ebay as well as several devices direct from voipsupply.com so it wouldn't sway me much if they were plugging their own company on this list, I already trust them. Never bought Polycom from them though, although I plan to in the near future. I have ordered from them too, and I was very happy with them. My only problem was with their deceitful wording of the post in question. -- Kristian Kielhofner ___ 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 -- MBM ___ 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] SIP-Asterisk-GnuGK-Cisco 5300
TO use * with a 5300 using SIP, just make a dial-peer and send the calls using dial([EMAIL PROTECTED]) should work, I'm using it go to and from. Actually its MUCH MORE Stable to use * in native SIP. The code for H323 it's not very updated. LTenorio -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Maynard Sent: Saturday, September 18, 2004 12:18 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP-Asterisk-GnuGK-Cisco 5300 Anyone has sample configuration files for this setup ? Start calls from a Sip client... and terminate them on a Cisco Gateway... going thru Asterisk and GnuGK? I have SIP clients working ok through asterisk ... I also have H323 Clients working great through GnuGK- Cisco Gateway What i've not been able to do is Terminate calls from SIP to the Cisco Gateway. Asterisk always get a security denial when trying to Register with GnuGK. Also... when i try to call from console i get a CallerNotRegistered error. I'd greatly appreciate anyone who can share Sample configuration files for Asterisk and GnuGK. Regards, Carlos. ___ 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] FC2 zaptel compile failure
Just take a look at the list, or search in google, you will find the answer. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff Borders Sent: Friday, September 17, 2004 4:41 PM To: Asterick Users Subject: [Asterisk-Users] FC2 zaptel compile failure I've got a fresh FC2 install and I'm trying to get the symlinks right according to the /usr/src/zaptel/README.Linux26 instructions. I've created two symlinks: /usr/src/linux-2.6 - /usr/src/linux-2.6.5-1.358 /lib/modules/linux-2.6 - /lib/modules/2.6.7-1.494.2.2 When I do a make linux26, I get a million warnings and errors with the result being: make[2]: *** [/usr/src/zaptel/zaptel.o] Error 1 make[1]: *** [/usr/src/zaptel] Error 2 make[1]: Leaving directory '/usr/src/linux-2.6.5-1.358' make: *** [linux26] Error 2 Am I looking at the right thing? Or do I have another problem? Jeff Borders [EMAIL PROTECTED] ___ 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] ERROR[16384]: chan_h323.c:1987 load_module: Gatekeeper registration failed
Actually, * it's not a GK, you should configure it as regular Terminal (Not a Gateway)in your GNUGK. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Maynard Sent: Thursday, September 16, 2004 10:56 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ERROR[16384]: chan_h323.c:1987 load_module: Gatekeeper registration failed I'm trying to configure Chan_H323 to register with GnuGK... without success... i've failed finding sample configurations. I'd greatly appreciate anyone who can provide sample config of H323.conf and gnugk.ini I am tyring to configure Asterisk as a neighbor in GnuGK. I'm always getting this error on Asterisk. ERROR[16384]: chan_h323.c:1987 load_module: Gatekeeper registration failed. *** And a SecurityDenial error on GnuGK. This is my H323.conf [general] port = 1720 bindaddr = 0.0.0.0 disallow=all allow=g729 allow=G723.1 allow=ulaw ; Allow codecs in order of preference allow=alaw gatekeeper = 66.118.228.198 context=h323 [1005] ; When this line and the context [1004] lines are set type=h323 ; the caller id 1004 is always sent. I don't know why. e164=011005 ; In case, this lines are not set, the GS phones receives context=default ; Error as the caller id, and the H323 phone receives ; asterisk as the caller-id [1004] type=h323 e164=011004 context=default [asterisk] type=h323 prefix=01 context=h323 This is my entry in GnuGK [RasSrv::Neighbors] asterisk=68.90.233.134;1720;01; I've configured other GKs using this Neighbors section and it doesn't require password. Best regards, Carlos Maynard Jr. [EMAIL PROTECTED] ___ 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] Fw: Asterisk R2 Signaling
I've seen a lot of times, people that try to get R2 MFC to *, most of them trying to use Dialogic Boards (BTW They 're Very expensive), none of them where succesfully, If you want to use PCI Cards on your server,why don´t u ask to your carrier to provide you E1/PRI? or better put a Gateway with E1s and SIP. I'm in Argentina also and I get PRI E1s from some of the Carriers From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sam NjengaSent: Wednesday, September 15, 2004 3:48 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Fw: Asterisk R2 Signaling Has anyone found a solution for asterisk and r2 signaling ? Steve Underwood had given some information saying he had a working asterisk working. I need it to work with Argentina R2 signaling Sam ___ 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] OH323 Trunking
OH323 registers itself as a Gateway, and the H323 channel as a terminal. Afaik there is no easy way to change it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas Sikkema Sent: Tuesday, September 14, 2004 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] OH323 Trunking [EMAIL PROTECTED] wrote: From what I can tell when I place an outbound call from Asterisk it always tries to use the first registered H323 alias... My dial plan in extensions just says Dial(OH323.) Unlees the gatekeeper rejects multiple calls from Asterisk, there's no need for multiple aliases. Just setting the correct caller ID should be enough. Does the H.323 channel register itself as an endpoint or as a gateway? I think the easiest solution would be to register itself as a gateway. -- Andreas SikkemaRits tele.com Scheepmakersstraat 11 3011 VH Rotterdam t: +31 (0)10 2245544f: +31 (0)10 2245540 ___ 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] OH323 Ignoring PROGRESS indication
Try, in the 53 (depends on the SW version u're using voice call send-alert Also if you're using PRI trunks you can use, in the Serial interface,. isdn send-alerting -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Maxim Litnitsky Sent: Wednesday, September 08, 2004 12:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] OH323 Ignoring PROGRESS indication Good time of day all! 1) I am trying to use as5300 and asterisk. As5300 sends calls to me. I get the following in * console: -- IAX2/magrathea/6 is making progress passing it to OH323/R27464 Sep 8 10:57:59 NOTICE[1140046640]: chan_oh323.c:1159 oh323_indicate: Ignoring PROGRESS indication. As5300 user does not hear anything, just silense instead of dial tones. My config is oh323.conf default config. 2) * logs CDR records with NO Answer and duration 0, but h323 shows duration 0. Why so? ___ 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] Cisco GW and DTMF problems
What version of IOS 're u using, and what's your dtmfmode in *? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Billy Huddleston Sent: Wednesday, September 08, 2004 6:29 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco GW and DTMF problems I'm stuck running a old copy of asterisk because something strange is going on in later versions of the CVS.. When I call in from a PSTN into my cisco 2610XM gateway which then routes the call to my asterisk box via sip.. Asterisk can no longer process DTMF tones generated by the calling party. This affects DISA, prompts and menus.. Has anyone else had this problem?? and use.. I DO have dtmf-relay rtp-nte toggled in my dial peer.. Thanks, Billy +--+ | Billy Huddleston Senior Systems Administrator | | Net-Express http://www.nxs.net | | 114 Sherway Rd. Voice: 865-691-2011 | | Knoxville, TN 37922 Fax: 865-691-9894 | | [EMAIL PROTECTED]| +--+ ___ 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] mpg123 - multiple instances, taxing CPU
Actually, I got almost the same issue (i´m not having such load), but I got defines 4 different moh and got 10 process (I check every time I restart * to kill all the mpg123 processes also. LTenorio -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman Sent: Friday, September 03, 2004 11:06 AM To: 'Matthew Boehm '; '[EMAIL PROTECTED] ' Subject: RE: [Asterisk-Users] mpg123 - multiple instances, taxing CPU check your musiconhold.conf, for each one you define you'l get an instance. -Original Message- From: Matthew Boehm To: [EMAIL PROTECTED] Sent: 03/09/04 15:04 Subject: [Asterisk-Users] mpg123 - multiple instances, taxing CPU Is there any reason why there should ever be more than 1 instance of mpg123 running on a * server? I just did an 'uptime' and noticed all 3 of my loads where over 3.00. 'top' showed 8 mpg123 processes all processing the same 3 songs (our background music). I tried to kill one of them but another one spawned in its place. Any ideas? Thanks, Matthew ___ 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 The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ 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] Vlan question
The other way, at least with Cisco Switches (I saw it also with 3com
switches some time ago), is to set every port with a secondary vlan.
What it does, it sent those 2 vlan u set to the port, it consume less
resources in your switch, and keep you far from confinguring vlan
filtering.
LTenorio
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Woods
Sent: Tuesday, August 17, 2004 11:43 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Vlan question
Your switch has to support vlan trunking, ie. 802.1q. You will have to
set the switch port to trunk, allowing the vlans that you want to be
avilable, then you will set the phone to trunk those two vlans,
assigning one to the pc port.
-Mark
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: 13 Aug 04, 2:38 PM
Subject: [Asterisk-Users] Vlan question
Hi,
I am setting up an Asterisk system with Cisco 7960 phones. I have a
PoE injector to insert between the patch panel and HP 2626 switch. I
plan to plug the users pc into the phone and the phone into the wall. I
would like the phones to have a seperate subnet from the phones for
performance reasons.
May be a silly question, but with the pc and phone sharing the same
switch port, how will it know to seperate the traffic and subnets?
Thanks
tm
ID:[{20040813172548.30403.1811044938-12.6.18.86}]
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RE: [Asterisk-Users] Avaya firmware
Just guessing, but 've you tried the to rename Sip_4602ap1_0.ebin to appsip.ebin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Johnson Sent: Tuesday, August 17, 2004 1:28 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Avaya firmware Aaron Johnson wrote: I attempted to update an Avaya 4602 phone to the latest SIP firmware and now the phone stops at the bootloader. It keeps requesting an appsip.ebin file from my HTTP server and is no longer checking my TFTP server for update files. Since no appsip.ebin file was included in the firmware update provided by Avaya, I have no idea where I am supposed to get this file. And also since it is no longer talking to my TFTP server, I can not go back. Does anyone have any suggestions? Thoroughly stumped, Aaron I did tell the phone to request the sipto3231_0.ebin file from the HTTP server and it does send a request. The http server responds by sending the file, and then the phone complains that the file is unavailable. This is mind-boggling. Any suggestions? ___ 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] DID Questions
Use sip debug, that should print all the trace info for the call, id you know the ip or the originating gateway/proxy, you cal also use sip debug IP -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Roberts Sent: Monday, August 16, 2004 6:38 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] DID Questions Is there anyway to test if this call is touching my servers? Everyone is telling me the DID is fine. But I can't confirm that for sure. And I don't want to go ahead and start making changes to my config when the DID isn't even working in the first place. ___ 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] Avaya and Asterisk
Yep, 4602 Works. That's what I said 'most of the phones' and ask for the model, I've tried 46XX without success, several times. LTenorio -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Elton Sent: Friday, August 13, 2004 12:38 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Avaya and Asterisk You can load the newly released Avaya firmware on the Avaya 4602's to register with Asterisk. I am doing this and it works. On Wed, 11 Aug 2004 11:53:07 -0700, Katerina Sadri [EMAIL PROTECTED] wrote: So far I have not found a way that I can register the Avaya phone with Asterisk. From what I have found so far is that Avaya phone needs the Avaya Media Server and Avaya Gateway. Looking at the h.323.conf (in Asterisk) and the file 46xxsettings.txt (avaya file located in tftpboot) there are no settings to make the phone initialize. I have sent an email to the Asterisk Users Mailing List to see if anyone has done it before. I will investigate some more and I hope that someone from the mailing list will answer. Katerina ___ 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] Avaya IP Phones and Asterisk
Be carefull, what's you phone type, most of the Avaya IP Phones use custom h.323 protocol. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Katerina Sadri Sent: Wednesday, August 11, 2004 4:06 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Avaya IP Phones and Asterisk I have added H.323 support for ASTERISK PBX using the OpenH323 library. I thought that since the Avaya IP phones are H.323 phones I will be able to make them register to Asterisk server. Does anyone know if this is possible ? Katerina ___ 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
Have u try disabling fast start?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Asterisk .
Sent: Friday, August 06, 2004 8:13 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] H323 Call Dropping
Please Help...
--- Asterisk . [EMAIL PROTECTED] wrote:
Hello,
--- Jeremy McNamara [EMAIL PROTECTED] wrote:
H323/${EXTEN} since your using a gatekeeper
Jeremy McNamara
Thanks to Jeremy, Sebastian and Thomas for the replies. But the calls
get dropped from the gatekeeper. Now i am trying only with g711
without using the gatekeeper, and sending the calls to Nextone. But no
success. Sometimes the call is accepted, but disconnects after 2
seconds. I have tried this with both chan_h323 and chan_oh323 and the
result is same. I am attaching herewith the log (chan_h323) of a call
which was dropped before answering.
Regards,
*CLI Urgent handler
7:57.232 ThreadID=0x0002e010 h323ep.cxx(1323) H323
Making call to:
h323:[EMAIL PROTECTED]
-- Called h323:[EMAIL PROTECTED] Urgent handler
7:57.236 H225 Caller:8132148 transports.cxx(1482)
H323TCP Started connection:
host=208.xxx.xxx.xxx:1720, if=208.xxx.xxx.253:36897, handle=36
7:57.237 H225 Caller:8132148 h323.cxx(3956) H245
Default
OnSelectLogicalChannels, FastStartDisabled
7:57.238 H225 Caller:8132148 h323.cxx(3144) H245
Started control channel
7:57.242 H225 Caller:8132148 h323.cxx(1591) H225
Reading PDUs:
callRef=25137
7:57.243 H225 Caller:8132148 h323.cxx(2095) H225
Set remote party name:
208.xxx.xxx.xxx
7:57.381 H225 Caller:8132148 h323.cxx(2095) H225
Set remote party name:
208.xxx.xxx.xxx
-- H323/208.xxx.xxx.xxx is ringing Urgent handler
8:09.772 H225 Caller:8132148 h323.cxx(2095) H225
Set remote party name:
208.xxx.xxx.xxx
8:09.773 H225 Caller:8132148 h323.cxx(2898) H225
Received connect PDU.
8:09.774 H225 Caller:8132148 transports.cxx(1482)
H323TCP Started connection:
host=208.xxx.xxx.xxx:37837, if=208.xxx.xxx.253:36898, handle=39
8:09.775 H245:814f110 h323.cxx(3144) H245
Started control channel
8:09.777 H225 Caller:8132148 h323.cxx(3144) H245
Started control channel
8:09.801 H245:814f110 h323neg.cxx(378) H245
MasterSlaveDetermination:
local is slave
8:09.802 H225 Caller:8132148 h323.cxx(1445) H225
Sending release complete
PDU: callRef=25137
8:09.805 H225 Caller:8132148 h323.cxx(1634) H225
Signal channel closed.
8:09.805 H245:814f110 h323.cxx(3195) H245
Read error: Bad file
descriptor
8:09.806 H245:814f110 h323.cxx(3207) H245
Control channel closed.
8:09.819 H323 Cleaner h323.cxx(1542) H323
Connection
ip$localhost/25137 terminated.
== No one is available to answer at this time Urgent handler
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RE: [Asterisk-Users] Large Enterprises using asterisk
I've been reading, learning from you guys and researching the various sites I've found dedicated to * for a while, trying to understand how it works and the best way to get the features work in our little environment (1 E1/8 POTS/30 VoIP phones, and a couple of Faxes/Modems/etc). I found a lot of useful information plus a lot of people working to make a better product with a lot of features that sometimes if you use commercial products require $40K/50K or more, VM it's a simple but a powerful one, IVR is great, and the way it manage the Dial Plan is another. That's why I think that * is a great product with a lot effort putted on it, with very useful features What I also saw in my little research that * in not suitable for large deployments like medium or large enterprises or sometimes even smalls ones with specifics needs, any of you could mention a lot of them, Call Centers, Banks, etc. and why not carriers for their own use. NOW, let me explain my POV. I worked for a large company (a Customer Management Company, sometimes wrong called a Call Center Company) a lot of years (like 10). I would disagree with Michael, it's not important to know Linux to make them use *, in those companies the people that take those decisions doesn't know even Windows, Solaris or IOS (those're just names) and more important they don't need to. They just know the brands (Avaya, Nortel, Cisco, Sun, HP, etc. just to mention some of them). They trust in people that make reports based on a couple of things, Brands, features, money and capacity for deployment. - Brand - The * brand it's Linux and GNU, which is good, I hear from a couple of companies that they 're starting to think to use it on some of their projects, just because the world it's spinning on it. First problem solved. - Money - * cost them nothing (which I think it's wrong) and it's not about TCO or ROI (Windows cost them a lot, not to mention Exchange which it's used on most of the large orgs. It's not quite cheap), I think that those orgs SHOULD have developers to contribute (yes contribute not steal for/from the community). Second problem almost, it will just take some time. - Features - * has a lot of features, not just the ones that came with the product, also the features that a lot of people help to develop, 4 reporting, dialers, protocols, etc. All of us with resources SHOULD put those resources (programmers, servers, time, Trunks) to help the community to grow (I'm trying to do that in my company) Third one is on the way, so don't worry on it. - Capacity for deployment - I know a lot of great products that just die because the brands don't saw the time coming (Novell with Netware it's and example, IMHO Netware, in their time, was a much better product that Windows NT Server) not even mention when Novell sold Unix). THE THING to change JIT could be anything, a way to sell, a way to marketing the product or the roadmap use to develop. Here is where I think * could be better, but the decision it's yours or better, OURS (users and developers), if we want to make * a product for the masses (just developers, techies???, and small companies) you're in the right way, if you want to make it a product for a Company or an org, you're not. DON'T GET ME WRONG, (I will be an * user and I will help the community to grow) and I'm not saying that * will be dead in the future. IMHO, * need a change in the architecture, in the way it's made, and in the way it works, not just more features or Bug solving. I'll be thinking, yes I also think, in the last weeks the way to help, I'm not a programmer so I could not develop, I don't even know Linux the way you do, so I could not help that way either, but I could help in other ways, That's why I take 2hs to write this email and I love to do it. Think on it. Think on an * system with servers for Trunks (IP, TDM, don't care), servers for core routing, servers for VM, servers used as IVRs, remember what those companies needs (redundancy, fault tolerance, load balancing, among other things) * don't have these features yet, think on systems with 600, 1K or even 3K users, think on systems that when a server crash (and believe, servers always crash) others take the place, with a little o even better with no downtime, 4 those users, trunks or systems. When that time comes, and trust me it could be done, * could be used in orgs. Sorry 4 my English, it's hard 2 explain in a foreign language what I try to. Just my 2c, Leandro -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanwar Ranbir Sandhu Sent: Thursday, July 22, 2004 7:12 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Large Enterprises using asterisk On Thu, 2004-07-22 at 10:16, Sunrise Ltd wrote: Michael Little wrote: Unfortunately, not everyone knows how to use Linux. While I don't disagree with your comments in general, I take issue with the
RE: [Asterisk-Users] SMDR/CDR - Asterisk integration
Seshu, I'm interested could u provide more info... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu Sent: Wednesday, July 14, 2004 11:02 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] SMDR/CDR - Asterisk integration Hi All, The CDR Tool in .PHP is working great. We have put this into production. Here is the Link: http://67.109.153.236/asterisk-stat/cdr.php If anyone is interested, I will generously contribute the code for your use. Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rich Allen Sent: Monday, July 12, 2004 11:16 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SMDR/CDR - Asterisk integration iH went to the link to take a look but admin/admin doesn't work - hcir On Jul 9, 2004, at 10:56 AM, San Singhania wrote: Hello everyone, I am developing an online SMDR / call log system for asterisk. This is going to take the form of an executable with embedded sql and webserver, pdf generation, excel generation, graphs. Actually, we have been selling this for a while now with great success and now I am starting work on the integration with Asterisk. Its a windows executbale and the executable is just about 1MB. If someone is interested, let me know. The online demo is at http://demo.callaccounting.ws . The username/password is admin and admin. To print out reports, just leave all the fields for the report selection blank. With regards, San ___ 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] Voice Numbers in Spain (SIP)
I'm looking for Spain Voice Numbers, anyone know a trustable company that provides them? TKX, Leandro ___ 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
