[Asterisk-Users] Asterisk Java-Call Problem
Hello,
i have some problem with the Asterisk and need a little bit help.
I have an dialplan like this:
[first]
.
..
...
exten = s,6,Read(Secret,,25)
exten = s,7,NoOp(**${Secret}**)
exten = s,8,Gotoif($[${Secret} = 999]?9:9)
exten = s,8,Wait(15)
When the user enter the secret key, my java- backend application validate the
entered user input.
If the user input is correct, i want that the user comes to the second context
controlled by my java application:
[second]
.
..
...
But how i can i send the next context information to an open phone line ?
I can't find any Java examples on the net.
If you have any idea / examples?
Thanks
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[Asterisk-Users] California DID Access
Hi, I am looking for a DID access provider in the west cost (not Iconnect, voiceglo, packet8 etc). What I need someone like NuFone, IAX and Toll/Tolfree multiple presentations possible. Any references appreciated. Thanks SW ___ 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] How to bind RTP when IP alias are configured
Hi Robert, Yes, this fixed the RTP issue for me. Do you need a bug note created on this ??? Cheers SW Date: Mon, 12 Jan 2004 14:03:14 -0800 (PST) Subject: Re: [Asterisk-Users] How to bind RTP when IP alias are configured From: Robert Hajime Lanning [EMAIL PROTECTED] To: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] quote who=SW Hi Folks, I have a situation where my Colo insists on a particular IP setup for my * server box. They allocate two blocks of IPs to my colo server. One set as my own (ex 20.20.20.20.4/30 - 4 ips) and the other as a transit lan (es 10.10.10.0/29). These are all public IP addresses and there is no NAT involved in. So essentially I have to set-up IP aliases in my Linux box as follows; Example: TRANSIT LAN: 10.10.10.0/29 CUSTOMER LAN: 20.20.20.20.4/30 RedHat LINUX FILE: /etc/sysconfig/network-scripts/ifcfg-eth0 DEVICE=eth0 IPADDR=20.20.20.20.4 NETMASK=255.255.255.255 ONBOOT=yes FILE: /etc/sysconfig/network-scripts/ifcfg-eth0:99 ## TRANSIT IP: DO NOT UNCONFIGURE ## DEVICE=eth0:99 IPADDR=10.10.10.4 NETMASK=255.255.255.248 NETWORK=10.10.10.0 BROADCAST=10.10.10.3 GATEWAY=10.10.10.1 ONBOOT=0 First of all. I can ping to customer lan and telnet to it, therefore IP routing (at least for unicast traffic) works fine. Now question arises when asterisk start to work on this box. Since the IP that I am supposed to use is 20.20.20.4, I set that as bindaddress in my sip.conf file. As far as SIP messages are concern * users that IP address, no problem. However for RTP stream * users 10.10.10.4 as it's source address. Because of this obviously calls will not go through asterisk, as the ip phone is expecting RTP packets from the SIP server which is bound to IP 20.20.20.4. Is there a way to tell * to use the same bind address in SIP.conf (h323.conf, iax.conf) for RTP ? I read rtp.conf file but that does not show any bind address. It seems like LINUX always select it's src address as the interface (alias) which has the gateway tied to it unless otherwise an application specifically asks Linux to use a particular ip address. rtp.c uses 0.0.0.0 (hardcoded, well kindof, the whole struct is initialized to zeros, so, it is just not set) Since, to get around NAT issues, I have a host route on my firewall (Linux IPTables), I have the same problem. In rtp.c - function ast_rtp_new()... rtp-us.sin_family = AF_INET; /* the next line was added to fix host route hack instead of NATing */ inet_aton(20.20.20.4,rtp-us.sin_addr); rtp-s = socket(AF_INET, SOCK_DGRAM, 0); Sorry, no context diffs. When I get around to adding an rtp.conf keyword, I will provide context diffs to bugtrack. -- END OF LINE -MCP ___ 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] cdr processing
Hi friends, Could some one recommand a good cdr processing software out there for post paid billing (invoicing, web-based payment processing) etc., Thanks a bunch. SW ___ 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] grandstream asterisk configuration
Hi, In my experience with GS phones, you need STUN support to make it work properly (behind NAT), otherwise you would need lot of trial end error to figure out how to do port forwarding. If you have STUN you wouldn't need to touch the Netgear (except for firewalls). If you can't run your own stun server (need two public IPs) then use one of many STUN servers out there on public internet. For an example enable NAT traversal on your GS phone and point the STUN server to one of these STUN servers larry.gloo.net or stun01.newkinetics.com. Then reboot the GS and see how it discover the NAT (top of the gs web GUI). If it is not a full cone or UDP blocked then you should be fine (Netgear is restricted cone). Cheers SW From: Chandra [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] grandstream asterisk configuration Date: Wed, 14 Jan 2004 19:35:48 +0545 Reply-To: [EMAIL PROTECTED] i have forwarded ports 5060 and 5000-5008 the ports used by sip and rtp to grandstream from my netgear. rtp.conf uses rtpstart 5000 and rtpend 5008. i have also opened all 5060, 5000-5008 ports in my firewall configuration. grandstream uses 5004 port for rtp. what am i missing here? please tell me. chandra ___ 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] Re: Proposed solution for exit code priority jumps
Hi John,
First, I have not much experience dealing with complex dial plans. But since
you asked, thought of some feedback.
In my opinion .conf files should be kept as simple as possible. It should
provide straight forward and simple manipulations to simple common
applications. If more complex manipulations are necessary, then those
scenarios could be built using scripts. Therefore, I think a structured
scripting implementation would be an option here. You may take a look at SER
and see how scriptic it's config files are. It is not simple but very
logical. So why patch what we have with another hack, which two will find
limitations. Instead, go for a more flexible script based dial plan.
Now If I have to choose one of two options you mentioned below, I would pick
the first one.
Cheers
SW
At 9:49 PM -0500 11/28/03, John Todd wrote:
Proposal for Alternate Error Handling Jumping
Why: I have written quite a bit into various extensions.conf files,
and I've started to find myself getting really, really frustrated
with the +101 and +51 and +blah format of error handling. I often
create very ugly and awkward dialing plans to handle jumps from (as
an example) multiple Dial statements which directly follow one
another. Hardcoding a Goto into each application seems to be a
method that, as Asterisk matures, should be left behind.
I have whined before about the lack of exit codes from many
applications (especially Dial) and perhaps there is some middle
ground. I have come up with two methods that might make the job of
the advanced administrator significantly easier, and dialplans more
compact. Additionally, logic for handling results of applications
would be visible in the same configuration line as the application,
instead of in a long chain of comparisons, or not at all, as is the
current case.
Both of these methods could be implemented (to the best of my
knowledge) without changing the way the application priority syntax
currently works, and are completely backwards compatible with
current methods. If this is not the case, I would appreciate
someone explaining how this could be better done, or why it should
not be done in the first place.
Alas, as with most of my proposals, I can only offer ideas and not
actually code them. Volunteers welcome.
Proposed Solution:
Alter the priority statement to take modifiers, if specified, so
that the three basic exit codes could be given different places to
land. In my example, exit-1 is the place where we should jump on a
-1 exit code, exit0 is where we go on a zero result, and exit1 is
error but continue in situations like Busy, and so on.
Applications like ENUMLookup, as an example, would have to document
two different error but continue codes, currently represented by
the +101 on no ENUM reply (turns into exit code 1) and +51 on TEL
(turns into exit code 2).
Syntax:
exten = extension,[priority[/exit-1[/exit0[/exit1[/...,Application
Exmaple:
exten = _87810.,1/h/2/4/10,EnumLookup(${EXTEN})
exten = _87810.,2,Dial(SIP/${ENUM})
exten = _87810.,3,Hangup
;
exten = _87810.,4,Answer
exten = _87810.,5,Playback(sorry-no-enum-information)
exten = _87810.,6,Hangup
;
exten = _87810.,10,Dial(Zap/g1/${ENUM})
exten = _87810.,11,Hangup
;
exten = h,1,Hangup
Alternate method (more complex):
Applications could exit with any number of codes, perhaps even
dynamic code results, and wildcards could be used to match on
priority jumping. This is a simpler method than setting an
arbitrary string as a result of an application and then using a
series of GotoIf statements to redirect call control. It is more
complex and completely encloses the purely ordinal solution I
describe as the first proposed solution. Each application might
have it's own list of exit codes which mean different things, or
dynamically exit with results that might allow the administrator to
take actions without having to set variables and create labyrinths
of GotoIf's upon an application's exit.
Syntax:
exten = extension,[priority[/pattern|priority[...]],Application
Example:
exten = 1234,1/_20.|cont/_40.|fail,Dial(SIP/1234)
exten = fail,1,Hangup
exten = cont,1,Playback(continuing_call)
In the above example, Dial would exit with something like 200
Completed and the priority would match against the 200 part of
that string and jump to extension cont. Similarly, 400 Failed
would jump to extension fail.
JT
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Re: [Asterisk-Users] Nufone.net net wackiness?
Well, I do not know what's going on with them, until late December I am waiting for an inbound DID number from NuFone :) Seems like telecom downturn is over and JJ has too many requests and couldn't handle it High time to hire some of those still looking for jobs. SW Message: 2 Subject: Re: [Asterisk-Users] Nufone.net net wackiness? From: Matteo Brancaleoni [EMAIL PROTECTED] To: [EMAIL PROTECTED] Organization: Espia - Emmegi Srl - Divisione IT Date: Tue, 13 Jan 2004 13:18:56 +0100 Reply-To: [EMAIL PROTECTED] only domain name screwed up. mmh.. my registrar allows me an autorenew for all domain names... pretty useful :) matteo. Il mar, 2004-01-13 alle 09:24, Chris Albertson ha scritto: Looks like they went off the air just after my PayPal payment was processed. I gues we wait a couple days to see if Nufone has gone belly up/bankrupt/gone or if this is just a domain name screw up. --- Steven Critchfield [EMAIL PROTECTED] wrote: On Tue, 2004-01-13 at 01:26, Brian Capouch wrote: I can't send mail to any addresses in nufone.net; they all get rejected by a spam blocker. And their website is gone, too!! The URL leads to a parking site. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK ___ 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] How to bind RTP when IP alias are configured
Hi Folks, I have a situation where my Colo insists on a particular IP setup for my * server box. They allocate two blocks of IPs to my colo server. One set as my own (ex 20.20.20.20.4/30 - 4 ips) and the other as a transit lan (es 10.10.10.0/29). These are all public IP addresses and there is no NAT involved in. So essentially I have to set-up IP aliases in my Linux box as follows; Example: TRANSIT LAN:10.10.10.0/29 CUSTOMER LAN: 20.20.20.20.4/30 RedHat LINUX FILE: /etc/sysconfig/network-scripts/ifcfg-eth0 DEVICE=eth0 IPADDR=20.20.20.20.4 NETMASK=255.255.255.255 ONBOOT=yes FILE: /etc/sysconfig/network-scripts/ifcfg-eth0:99 ## TRANSIT IP: DO NOT UNCONFIGURE ## DEVICE=eth0:99 IPADDR=10.10.10.4 NETMASK=255.255.255.248 NETWORK=10.10.10.0 BROADCAST=10.10.10.3 GATEWAY=10.10.10.1 ONBOOT=0 First of all. I can ping to customer lan and telnet to it, therefore IP routing (at least for unicast traffic) works fine. Now question arises when asterisk start to work on this box. Since the IP that I am supposed to use is 20.20.20.4, I set that as bindaddress in my sip.conf file. As far as SIP messages are concern * users that IP address, no problem. However for RTP stream * users 10.10.10.4 as it's source address. Because of this obviously calls will not go through asterisk, as the ip phone is expecting RTP packets from the SIP server which is bound to IP 20.20.20.4. Is there a way to tell * to use the same bind address in SIP.conf (h323.conf, iax.conf) for RTP ? I read rtp.conf file but that does not show any bind address. It seems like LINUX always select it's src address as the interface (alias) which has the gateway tied to it unless otherwise an application specifically asks Linux to use a particular ip address. Cheers SW ___ 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] FW: How to bind RTP when IP alias are configured
Hi Folks, I have a situation where my Colo insists on a particular IP setup for my * server box. They allocate two blocks of IPs to my colo server. One set as my own (ex 20.20.20.20.4/30 - 4 ips) and the other as a transit lan (es 10.10.10.0/29). These are all public IP addresses and there is no NAT involved in. So essentially I have to set-up IP aliases in my Linux box as follows; Example: TRANSIT LAN: 10.10.10.0/29 CUSTOMER LAN: 20.20.20.20.4/30 RedHat LINUX FILE: /etc/sysconfig/network-scripts/ifcfg-eth0 DEVICE=eth0 IPADDR=20.20.20.20.4 NETMASK=255.255.255.255 ONBOOT=yes FILE: /etc/sysconfig/network-scripts/ifcfg-eth0:99 ## TRANSIT IP: DO NOT UNCONFIGURE ## DEVICE=eth0:99 IPADDR=10.10.10.4 NETMASK=255.255.255.248 NETWORK=10.10.10.0 BROADCAST=10.10.10.3 GATEWAY=10.10.10.1 ONBOOT=0 First of all. I can ping to customer lan and telnet to it, therefore IP routing (at least for unicast traffic) works fine. Now question arises when asterisk start to work on this box. Since the IP that I am supposed to use is 20.20.20.4, I set that as bindaddress in my sip.conf file. As far as SIP messages are concern * users that IP address, no problem. However for RTP stream * users 10.10.10.4 as it's source address. Because of this obviously calls will not go through asterisk, as the ip phone is expecting RTP packets from the SIP server which is bound to IP 20.20.20.4. Is there a way to tell * to use the same bind address in SIP.conf (h323.conf, iax.conf) for RTP ? I read rtp.conf file but that does not show any bind address. It seems like LINUX always select it's src address as the interface (alias) which has the gateway tied to it unless otherwise an application specifically asks Linux to use a particular ip address. Cheers SW ___ 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] 2nd call leg status?
Hi Folks,
Wonder whether this question found an answer ?
I too have a similar question that I can't find an answer so far.
Let me first share my dial plan;
exten = _011.,1,Authenticate(/etc/asterisk/auth.txt |a)
exten = _011.,2,Playback(Pls-wait-while-I-connect)
exten = _011.,3,Absolutetimeout(3600)
exten = _011.,4,Dial(H323/${EXTEN:[EMAIL PROTECTED],70)
exten = _011.,5,NoCDR() =if no answer cdr is not written
exten = _011.,6,Busy
exten = _011.,105,NoCDR() if called party is busy cdr not wirtten
exten = _011.,106,Busy
exten = i,1,NoCDR() == if authentication failed cdr not written
exten = i,2,Hangup
Here are my observations
(a) Since Authenticate function is present in my dial paln, disposition
fieled in cdr always show Answered, so with that I can't figure out whether
H323 leg is successfully answered or not.
(b) If the H323 g/w sends the busy signal then CDR is not written, If the
g/w rings and timed out then again CDR is not written (as expected we have
priorities set for extensions)
(c) Now if the called party is ringing and originating party just hang-up, A
CDR is written. I have no way to differentiate that with a very short answer
call.
I think this behavior is a incorrect.
If * answered a call and show up disposition as Answered, then that call is
a completed call. So there should be one record in the CDR.
Then for the second leg there should be another record, as now * is
originating a call again and expecting other side(in this case the h323 g/w)
to respond.
If the both legs are considered as a single call, then cdr should show the
disposition of the final end point.
Please show me if there is a way that I could generate two records in CDR
for this kind of a call, or any other solution for this problem.
Cheers
SW
Message: 2
Date: Thu, 8 Jan 2004 17:27:07 + (GMT)
From: Stephen J. Wilcox [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] 2nd call leg status?
Reply-To: [EMAIL PROTECTED]
Hi,
okay heres what I want to do .. simple ivr, we take a call, answer it, play
a
menu, dial out based on options. No problems so far.
The CDR always shows the call as answered as I answer the 1st leg to play
the
prompts, I am actually more interested in if the 2nd leg - the outbound
part -
has been answered or not before the call is hungup. How can I get this and
record the information in the CDR?
Steve
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RE : [Asterisk-Users] 2nd call leg status?
Thanks Freddi, Cool, it works, infact better yet ResetCDR(). ResetCDR() show just one record in cdr, ResetCDR(w) will crete two records with the one for first call leg, which I do not want :). Cheers SW Date: Sun, 11 Jan 2004 17:27:08 +0100 From: Freddi Hansen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE : [Asterisk-Users] 2nd call leg status? Reply-To: [EMAIL PROTECTED] Hi, I did have the same problem. I simply issue a ResetCDR(w) as the last thing before using the Dial application. This will reset the 'answered' flag and the CDR you get from the Dial will contain the correct value. I do later do a small of backend processing to get the correct A-line time related to the call. There may be smarter way of doing this but this quick hack works fine for me Freddi ___ 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] Billing system experiences with Advanced Communications?
Message: 2 John, I am also considering this system. One of my friend has been evaluating this for couple of months now. He is still not confident though. However he said the features are promising. I will ask for more information. This is a company in Pakistan. You call their number in LA, the guy's will still answer from Pakistan. That may be why it is not so expensive. Hope to hear from your experience too. Cheers SW Date: Thu, 8 Jan 2004 10:21:06 -0500 To: [EMAIL PROTECTED] From: John Todd [EMAIL PROTECTED] Subject: [Asterisk-Users] Billing system experiences with Advanced Communications? Reply-To: [EMAIL PROTECTED] I ran across this URL the other day: http://www.advancedvoip.com/ It seems that they have a low-priced solution for billing, which seems to have some interesting features (at least, via the canned demo they provide.) It handles CDRs and also (sorry, Jeremy) Radius, and seems to have a lot of features. I'll be calling them later today to discuss their product, if only out of curiosity at their low prices and reasonably full-featured demo. The system seems very much geared to H.323, and it is not obvious what the back end is, so I am somewhat suspicious of functionality. Pricing seems reasonable - they charge by the port. For an 8 port system, it would be $960 (!!!) which probably translates into $1720 for a single 4-port PRI card. No recurring charges. It sounds almost too good to be true. I have had enough experience with billing systems to know at least that nothing ever is as it seems in the demo. Has anyone who is a commercial service provider on the list implemented this software? If so, what have your experiences been? JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: 2.4 Kernel and Hyperthreading (was Re: [Asterisk-Users] P4 processor
but if I disable hyperthreading (noht on the kernel line in lilo/grub) Could the same result could be obtained if hyperthreading is disabled from the BIOS ? Supermicro Motherboards (most Phoneix based) have this option Also, would we see same pops and clicks in a dual CPU box (Zeon) then ? SW Message: 1 Date: Wed, 07 Jan 2004 14:32:16 -0500 From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: 2.4 Kernel and Hyperthreading (was Re: [Asterisk-Users] P4 processor with Hyperthreading and Asterisk) Reply-To: [EMAIL PROTECTED] mattf wrote: Is Asterisk's high memory usage canceling out most of the performance gained by using HT? In my experience, I've found that hyperthreading is a major problem on Linux Kernel 2.4. If I turn on hyperthreading and start to load up an Asterisk box we get ratty sounding audio along with pops, clicks and ticks, but if I disable hyperthreading (noht on the kernel line in lilo/grub) and all of those anomalies go away and the box scales a whole lot higher. The problem is that the 2.4 Kernel sees the one hyperthreaded processor as two totally separate processors, so the 2.4 kernel tends to mess up, badly, when scheduling the processing. This problem was delt with in the 2.5 kernel and i've been told by quite a few trusted people that it works flawlessly, but i'm not about to run a unstable kernel (or even anything less than like 2.6.14) on any of my production boxes. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Codec Negotiation Does not seem to work as expected ?? Help Please !!
Try using canreinvite=yes in all three contexts. Would that screw-up ATA, I do not know, cause I have no Cisco's ? SW Message: 5 Date: Mon, 05 Jan 2004 02:29:49 -0500 From: SamW [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Codec Negotiation Does not seem to work as expected ?? Help Please !! Reply-To: [EMAIL PROTECTED] Hello, I have been trying to get my coders to work without a conversion. I have read all the available asterisk documentation and support groups without any luck. Here is my issue. (Please feel free to ask questions if you do not understand what I am talking about.) I am using Cisco ATA-186 set to g729 codec. (But it will switch to g711 if sip-server request g711) I have 2 SIP-services to which I have to deliver the call in 2 coder formats. Lets call 2 sip-providers, SIP-A and SIP-B. SIP-A accept g729 and g711, SIP-B only accept g711. I do not have any g729 licence, but I believe the * should negotiate to have the correct passthrough coders as ATA is capable of both coders. (I think even if you have the licenses, * should try avoid codec-conversions when ever it can) Here is my settings in sip.conf. I will only list the required codec related lines, for easy understanding, [general] disallow=all allow=g729 allow=ulaw allow=alaw register = [EMAIL PROTECTED] register = [EMAIL PROTECTED] [sip-a] disallow=all allow=ulaw [sip-b] ... disallow=all allow=g729 [ATA] . canreinvite=no ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE:[Asterisk-Users] Grandstream Handytone 286 RTP Problems
Hi Mike, I have handytones working OK with *. My username and name of the context set to same. Try this; [131] username=131 type=friend host=dynamic disallow=all allow=alaw allow=ulaw However I am wondering why you get destination unreacherbale from the handytone. This is nothing to do with SIP negotiation. So you might want to go look at the RTP trace and see whether those ports are blocked in your linux box. SIPURA might be using different set of ports. Also you should upgrade the handytones to a better code. I have 1.0.4.26, which is known to be pretty stable. If u want the code, search the mailing list, I remember in December someone posted where to download the code. SW Message: 1 From: Mike Machado [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Sun, 04 Jan 2004 20:16:31 -0800 Subject: [Asterisk-Users] Grandstream Handytone 286 RTP Problems Reply-To: [EMAIL PROTECTED] I am trying to get the handytone 286 to make a very simple call to * and having problems. It registers with * just fine, but when I place a call (to echo test, for example), the RTP stream seems to have problems opening. Here is there error I get in *: WARNING[98311]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 0 (Response) When doing traces with ethereal, I see successful SIP and SDP handshakes, but when * sends handytone RTP packets, I see a ICMP Port Unreachable messages sent from Handytone to * regarding the UDP RTP packet. * then gives up and I see a BYE from *, which handytone acks. Handytone config is default except obvious SIP registration parameters. I also have a Sipura SPA2000 and everything works perfect for that one, same extension and everything (not at same time of course). sip.conf entry: disallow=all; Disallow all codecs allow=ilbc allow=ulaw ; Allow codecs in order of preference [131] type=friend host=dynamic reinvite=no canreinvite=no qualify=300 callerid=handytone 131 mailbox=131 nat=0 Handytone info: Software Version:Program--1.0.4.17Bootloader--1.0.0.11 HTML--1.0.0.19 Both on same subnet, no NAT. I have two Handytones, both exhibit same symptoms. Anyone else have this problem? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie - getting two local phones tocommunicate would be a good start :)
John, Obviousely, this would not work. Look at my example before; [5702] == type=friend username=5702 == context=internal dtmfmode=info username and context should match. Better get it working in a simple LAN first, why NAT, why voicemail .. Go to basics :), SW From: John Coll [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Newbie - getting two local phones tocommunicate would be a good start :) Date: Sat, 3 Jan 2004 16:55:45 - Reply-To: [EMAIL PROTECTED] ; liza:/etc/asterisk/sip.conf ; [general] port = 5060 bindaddr = 0.0.0.0 externip = 10.0.1.198 [5702] type=friend host=dynamic context=johnhome reinvite=no canreinvite=no qualify=300 callerid=John workroom #1 5702 mailbox=5702 nat=yes [5703] type=friend host=dynamic context=johnhome reinvite=no canreinvite=no qualify=300 callerid=John workroom #2 5703 mailbox=5703 nat=yes ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie - getting two local phones to communicate would be a good start :)
Hi John,
If your effort is to make calls between two GS phones via *, here is what
you need.
You need all three devices in the same LAN, so set both phones and * to
10.0.1.98/24.
After that from your asterisk Linux box ping both phones. If that is
successful you know your layer 1,2 and 3 are ok. Disable all fire-walls,
iptables, ipchains in Linux box.
Now in * you need two files in /etc/asterisk
sip.conf and extensions.conf.
rename or delete both those existing files.
Here are the minimum you probably need in these two files.
sip.conf :
[general]
port=5060
allow=all
maxexpirey=180
defaultexpirey=160
[5702]
type=friend
username=5702
context=internal
dtmfmode=info
[5703]
type=friend
username=5703
context=internal
dtmfmode=info
And extensions.conf
[internal]
exten = _57XX,1,Dial(SIP/${EXTEN})
Save both files and issue command reload from * CLI.
Now you should be able to call from one phone to another.
while making calls enable sip debug and study the messages going in and out.
Also if you have ethereal fire that up and capture SIP packets. and see how
the SIP negotiation goes on. This will help you when you start moving to
fwd, IAXTEL etc. etc.
good luck.
SW
From: John Coll [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Date: Fri, 2 Jan 2004 22:57:28 -
Subject: [Asterisk-Users] Newbie - getting two local phones to communicate
would be a good start :)
Reply-To: [EMAIL PROTECTED]
Hi
This is hard work :) I have read the Asterisk Handbook, BudgeTone User
Manual, Andy Powell's useful notes, Zac Sprackett's Asterisk Resource Pages
and more.
I am not a linux newbie but am new to Asterisk. I have failed to find any
docs that explain how to get a very very simple, minimal, system up and I am
trying to get the following to work:
2 BudgePhone 102D connected on a LAN to a linux RH9 server running Asterisk
IP 10.0.1.198 - I just want to be able to dial from one phone and talk to
the other :) I have another phone connected to FWD sucesfully and the LAN is
NATed at the PC that is acting as the Asteriski server and firewall. But for
now its just two phones on a LAN - I'll conquer FWD and IAX later
The extensions are 5702 and 5703. I can dial direct from one phone to the
other (not using Asterisk) and the other one rings and answers fine with a
voice path.
When I dial 5703 from 5702 (thus via asterisk), 5703 rings. If I take it
off hook it stops ringing but I can still hear ringing on 5702. After a few
seconds I get the rapid-beep tone on both phones. No voice.
I get this from asterisk CLI
*CLI
*CLI
-- Executing Macro(SIP/5702-a5be, exten-aa|5703) in new stack
-- Executing SetMusicOnHold(SIP/5702-a5be, random) in new stack
-- Executing Macro(SIP/5702-a5be, dial|20|tr|5703) in new stack
-- Executing AGI(SIP/5702-a5be, dialparties.agi) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Caller ID name is 'John workroom #1' number is '5702'
-- dialparties.agi: Added extension 5703 to extension map
-- dialparties.agi: Extension 5703 cf is disabled
-- dialparties.agi: Extension 5703 do not disturb is disabled
-- dialparties.agi: DbSet CALLTRACE/5703 to 5702
dialparties.agi: About to execute Dial(SIP/5703|20|tr)
-- AGI Script Executing Application: (Dial) Options: (SIP/5703|20|tr)
-- Called 5703
-- SIP/5703-5fdc is ringing
-- SIP/5703-5fdc answered SIP/5702-a5be
-- Attempting native bridge of SIP/5702-a5be and SIP/5703-5fdc
WARNING[81926]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries
exceeded on call [EMAIL PROTECTED] for seqno
36119 (Response)
== Spawn extension (macro-dial, s, 1) exited non-zero on 'SIP/5702-a5be'
in macro 'dial'
== Spawn extension (macro-exten-aa, s, 2) exited non-zero on
'SIP/5702-a5be' in macro 'exten-aa'
== Spawn extension (johnhome, s, 1) exited non-zero on 'SIP/5702-a5be'
*CLI
*CLI
I've turned on SIP debug but can not see any errors reported. This look like
the moment of failure:
Retransmitting #5 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.202
From: John Coll 5702
sip:[EMAIL PROTECTED];user=phone;tag=bfbd6f17-1d79-ed6b-1710-239de5724559
To: sip:[EMAIL PROTECTED];user=phone;tag=as3835ce1f
Call-ID: [EMAIL PROTECTED]
CSeq: 28108 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Type: application/sdp
Content-Length: 176
v=0
o=root 27210 27211 IN IP4 10.0.1.198
s=session
c=IN IP4 10.0.1.198
t=0 0
m=audio 18922 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
to 10.0.1.202:5060
WARNING[81926]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries
exceeded on call [EMAIL PROTECTED] for seqno
28108 (Response)
___
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[Asterisk-Users] asterisk gateway to other gateways
I was wondering why bother with this dialer thing at all :) When you implement a new thing, it is a good idea to go back to basics and think what you want to do. You have multiple voip carriers and you have bunch of users. Based on the number dialed you want to pick a carrier and dial out. May be authenticate a certain caller for certain path. Then end of the day look for CDR's. You do not need to set your call setup this fragmented. How about this; (a) User dials a international number (b) * looks at the CallerID and Destination Number (at one database lookup) (c) This caller is not allowed to dial this destination, so play a message for that extent (d) If caller is allowed, then * finds the least cost root and dial through that path and connect the call. * can do all these. So I would first sit and right down what I want my system to do (in my users perspective). Then try to find a way to implement. I wouldn't try to replace a box with another box. my two cents SW Message: 4 From: lito lampitoc [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Fri, 02 Jan 2004 01:04:40 +0800 Subject: [Asterisk-Users] asterisk gateway to other gateways Reply-To: [EMAIL PROTECTED] Though I've had implementations of Asterisk, I havent encountered this one yet, so i'd like to seek your advise if this possible. I would want asterisk to be stand between the dialer the destination. The dialer will now dial asterisk access number. Asterisk will acknowledge user by using CallerID and check against its database for authentication and then sends out a DTMF A tone for ½ second to enable the dialer to send the whole overseas digit. Assume the caller is not in database, asterik could give user a busy tone, IVR or just leave it and sends out a DTMF A tone anyway. Once the overseas digit are sent from dialer to asterik, asterik will then decide which telco/carrier/Voip to send the traffic to using LCR. Please note that we need to assign at least 5-10 telco/carrier/Voip access number for backup purposes. Once the least cost destination is selected by asterik, asterik will pick up the PRI line and dial a local access number and waits for a DTMF A tone. Once the A tone is heard from telco/carrier/Voip, it will send the overseas digit which was sent by the dialer earlier on. Also, can asterik sends out a musical tone or IVR while connecting to other telco to advice user that the call is connecting, else it would be dead air from there on. The whole process takes less than 5 seconds while the user stays on the line for this whole thing to happen. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New to asterisk? RUN... don't walk.
Hello, I am not a veteran here, but would like to share my thoughts on this subject. True, * is opensource and freely available, but it is not a computer program that you download and run. It is a very versatile telecommunication product you would otherwise pay at least 100 K to buy from a telecom vendor, if not more based on modules and usage, license hash-codes etc. Even to try * one would need some pre requisite knowledge in telecom, if not many years in the field. I work for a large telecom company and my specialty is voice over broadband (or xDSL). I worked with asterisk for couple of months now and I am amazed to see areas of telecom that * touch upon with. Starting from Linux, to SIP, H323, DSL technologies (PPP, PPPoE, PPPoA, DHCP, NAT), Call routing(Dial Plan), IVR, Transcoding, STUN are few areas that one would have to master even thinking about *. True one would know the syntax, and howtos etc, but also would have to have the ability to troubleshoot. For last two-three months in this list, I have not seen any newbi posting a sip trace (from a ethereal or a TCP dump) and asking a question about it. I have seen many question for instance, asking syntax of h.323 dial, but never seen a question asked on a h323 trace. I think, having * openly available is like keeping an airplane openly available in a airfield, so that anybody can try flying. Tell me how many of us would go try and fly that airplane if we do not know how to fly :) Point that I want to make here is simple, please try to understand what * is all about. If you like it's features and would like it to run in a production environment try to get some professional help. If you are learning these technologies for fun then get educated, use tools available to troubleshoot. Hooking up couple of phones and making a call is far from knowing *. Asterisk is a great product (thanks Mark and many others) and if you know what you are doing, you can do wonders with it. Don't put it down, because you do not have the background to understand it or work with it. Cheers SW Message: 4 Date: Wed, 31 Dec 2003 12:37:24 -0800 (PST) From: Me [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] New to asterisk? RUN... don't walk. Reply-To: [EMAIL PROTECTED] As a newcomer to Asterisk, you will not be welcomed with open arms. First, you will find almost no documentation on it's features. Second, if you try to ask questions, you will be flamed and pointed to worthless how-tos and 'the wiki'. These worthless documents can only be useful for explaining how things work to those already in-the-know. Lastly, Asterisk is so bug ridden, expect frequent segmentation faults. With a community so 'anti-n00b', don't expect your problems to be fixed anytime soon. RUN!!! Don't walk... away from Aterisk. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bandwidth requirement
Hi Folks, have a question, on bandwidth. I want to run an asterisk server SIP to H323, g729. Calls arrive on sip/iax go to IVR get authenticated and egress through h323. So G729 license is only used during IVR and then it is pass through. I am collocating this server. Colo offer a monthly bandwidth quota. Lets say I want to do 100K minutes per month of VoIP calling at the beginning. What would be the bandwidth usage per month ? I calculate this way, G.729 call is about 20K, but since call arrive on SIP and leave through h323 there are sessions with SIP server as well as H323 g/w. So a call is actually 40K. Since colo calculate bw in both upstream and downstream, total b/w is 2 x 40k = 80k. So 100K minutes is 100k x 80k = 80,00,000 kbps/month = 8Gig/month. Would my line of thinking correct ? As always for experts on this list to comment. Thanks SW ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] when * start at bootup chan_h323 fails to load
Hi Gurus I am trying to make asterisk load as a linux servics at boot time. I tried both methods; (a) /etc/init.d/asterisk (b) /etc/rc.d/rc.local But * failed to start. What is interesting is the message log (attached below), in either case problem is with chan_h323.so. Which is failing to load. Once the box is booted up I can start * no problems, I can run same asterisk script I used in (a) above and have no problems. Chan_h323 has no complains. So, what could be the difference at boot-time and when I manually run the same script later on ? Here is the log; Dec 23 23:33:50 WARNING[1074494176]: File cdr_addon_mysql.c, Line 258 (my_load_module): MySQL database sock file not specified. Using default Dec 23 23:33:50 WARNING[1074494176]: File chan_iax2.c, Line 5466 (set_config): Ignoring port for now Dec 23 23:33:51 WARNING[1142106560]: File chan_oss.c, Line 238 (sound_thread): Read error on sound device: Resource temporarily unavailable Dec 23 23:33:51 WARNING[1074494176]: File chan_zap.c, Line 7341 (setup_zap): Ignoring rxwink Dec 23 23:33:52 WARNING[1074494176]: File loader.c, Line 239 (ast_load_resource): libpt_linux_x86_r.so.1.5.2: cannot open shared object file: No such file or directory Dec 23 23:33:52 WARNING[1074494176]: File loader.c, Line 407 (load_modules): Loading module chan_h323.so failed! Any help greatly appreciated !!! SW ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] when * start at bootup chan_h323 fails to load
Jeremy, Ok, that worked. Thanks for your help, really appreciate it. Let me copy this to the list, someone will find it useful. So, If you want to run * at bootup, and you have chan_h323, (a) then you should modyfy init.asterisk script with the path variables (shown below) and copy it to /etc/init.d, rename to asterisk (or anything) (b) Then do chkconfig --add asterisk and (c) chkconfig asterisk on (d) Now reboot and asterisk will start as a service Merry Christmas. Cheers SW -Original Message- From: Jeremy McNamara [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 24, 2003 2:25 PM To: SW Subject: Re: [Asterisk-Users] when * start at bootup chan_h323 fails to load You answered your own question here. Your startup environment does not use /etc/profile, so you have to copy those same directives into the asterisk startup script, so its environment is properly setup. Jeremy SW wrote: Hi Jeremy, I did read the README. Infact I knew you would love to RTFM :). Actually, I created the environment for BASH, exactly the way you asked to do. The question here is; chan_h323 get started, when I login as root and when I run the same script that I have in /etc/init.d. But it complaints when it is run at the boot time(so the path is good for user root, but not good when it is started at boot time). So, I must be doing something wrong in setting the environment, which seems only effective when logged in as root. I am running rh 9, and I put those path variables in /etc/profile. Here is my /etc/profile HOSTNAME=`/bin/hostname` HISTSIZE=1000 if [ -z $INPUTRC -a ! -f $HOME/.inputrc ]; then INPUTRC=/etc/inputrc fi export PATH USER LOGNAME MAIL HOSTNAME HISTSIZE INPUTRC for i in /etc/profile.d/*.sh ; do if [ -r $i ]; then . $i fi done unset i PWLIBDIR=/root/pwlib export PWLIBDIR OPENH323DIR=/root/openh323 export OPENH323DIR LD_LIBRARY_PATH=$PWLIBDIR/lib:$OPENH323DIR/lib export LD_LIBRARY_PATH also, echo $LD_LIBRARY_PATH gives me what it is. [EMAIL PROTECTED] sath]$ echo $LD_LIBRARY_PATH /root/pwlib/lib:/root/openh323/lib Is there any other log where we can take a closer look ? Would a complete clean and make of pwlib and openh323 would help? Things work fine, as far as call processing is concern, so I am reluctant to mess the installation again. Cheers SW Date: Wed, 24 Dec 2003 10:15:04 -0500 From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] when * start at bootup chan_h323 fails to load Reply-To: [EMAIL PROTECTED] SW wrote: (ast_load_resource): libpt_linux_x86_r.so.1.5.2: cannot open shared object file: No such file or directory Dec 23 23:33:52 WARNING[1074494176]: File loader.c, Line 407 (load_modules): Loading module chan_h323.so failed! RTFM cat /path/to/asterisk/channels/h323/README Jeremy McNamara libpt_linux_x86_r.so.1: cannot open shared object file: No such file or directory You have not set the LD_LIBRARY_PATH environment variable. Example environment for sh/bash: PWLIBDIR=$HOME/pwlib export PWLIBDIR OPENH323DIR=$HOME/openh323 export OPENH323DIR LD_LIBRARY_PATH=$PWLIBDIR/lib:$OPENH323DIR/lib export LD_LIBRARY_PATH We recomend puting the above directives into your /etc/profile so you do not have to remember to export those values every time you want to recompile. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Auto Starting Asterisk
Hi Chris, In this situation, how do I modprobe ztdumy before * get started ? SW Message: 6 Date: Tue, 23 Dec 2003 09:33:07 -0800 (PST) From: Chris Albertson [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Auto Starting Asterisk To: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Look in the directory /etc/init.d (/etc/rc.d/init.d on some systems) You put a script in there called asterisk. There is a sample called asterisk.init in the source. copy it to /etc/init.d/asterisk You may want to study the other files in /etc/init.d to see how they work. Next read the chkconfig man page and see way you'd want to type chkconfig --add asterisk; chkconfig asterisl on Finally to start asterisk you can type ./asterisk start You may also want to re-boot the computer to verify that asterisk does start automatically ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] x100P incoming
Exten = s,1,Wait(1000) This will make * not answering the call, but still you would see notices coming on your screen and also an entry in CDR. immediate=no Will try this out SW From: Tilghman Lesher [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] x100P incoming Date: Sat, 20 Dec 2003 11:10:26 -0600 Reply-To: [EMAIL PROTECTED] On Friday 19 December 2003 08:12, David Gomillion wrote: How do I make x100P does not answer incoming calls ? The only thing that springs to mind is that you create an incoming context, and have an extension like: Exten = s,1,Wait(1000) Dunno if it will work or not, but that's the only thing that springs to mind. What about setting in zapata.conf: immediate=no before the channel declaration for the X100P. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Re: [Asterisk-Users] x100P incoming
immediate=no did not work either, so at least to get * not write to CDR, I will have to use command noCDR(). My extension context is as follows [x100pincoming] exten = s,1,Ringing exten = s,2,Wait(40) exten = s,3,NoCDR() It still write the CDR in MySQL, any idea how to get rid of that ? SW -Original Message- From: SW [mailto:[EMAIL PROTECTED] Sent: Saturday, December 20, 2003 10:08 AM To: [EMAIL PROTECTED] Digium. Com Subject: Re: [Asterisk-Users] x100P incoming Exten = s,1,Wait(1000) This will make * not answering the call, but still you would see notices coming on your screen and also an entry in CDR. immediate=no Will try this out SW From: Tilghman Lesher [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] x100P incoming Date: Sat, 20 Dec 2003 11:10:26 -0600 Reply-To: [EMAIL PROTECTED] On Friday 19 December 2003 08:12, David Gomillion wrote: How do I make x100P does not answer incoming calls ? The only thing that springs to mind is that you create an incoming context, and have an extension like: Exten = s,1,Wait(1000) Dunno if it will work or not, but that's the only thing that springs to mind. What about setting in zapata.conf: immediate=no before the channel declaration for the X100P. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Land line vs. VoIP provider.
Hi Stephen, Interesting 6) When, and if, the quality/reliability improves sufficiently, a DID line in the area code of your choice, which provides 6 simultaneous call presentations for $7.99/month, will beat any land line hands down. I did not know that one connection can have many simultaneous call presentations. I just tried with my iconnect account, So while I was calling in to an * extension, I tried calling the same iconnect number again. It didn't give me an engage tone, infact it came to the * as well. Six simultaneous presentations you mentioned here is for VoicePulse, right ? Do you know the limitation for Iconnect ? I can't use VoicePulse because I live in California, and need California numbers. Thanks SW ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Land line vs. VoIP provider.
As far as I know, iconnect explicitly disallows multiple call presentations. The iconnect thing was discussed on the list a month or two back. I red that discussion, it was more on multiple outgoing calls. I noticed with Iconnect, the sip invite message always comes as the [EMAIL PROTECTED] So if I use two PSTN numbers to call same Iconnect provided number, it arrives at asterisk as two different calls. Iconnect seems like completely depends on the SIP response from * or any SIP end point to figure out the call status (busy or on call). That is why I think, it is IMPOSSIBLE to set the * to receive a call coming from Iconnect to a particular context, other than general section, in it's sip.conf. (I didn't hear from any one who got this working) I think this is how, fwd and others are offering access numbers in many US States. Using a single number many callers can get to their (fwd) network. This is my observation, please correct me if I am wrong ? SW -Original Message- From: Stephen R. Besch [mailto:[EMAIL PROTECTED] Sent: Friday, December 19, 2003 2:29 PM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: Re: Land line vs. VoIP provider. SW wrote: Hi Stephen, Interesting 6) When, and if, the quality/reliability improves sufficiently, a DID line in the area code of your choice, which provides 6 simultaneous call presentations for $7.99/month, will beat any land line hands down. I did not know that one connection can have many simultaneous call presentations. I just tried with my iconnect account, So while I was calling in to an * extension, I tried calling the same iconnect number again. It didn't give me an engage tone, infact it came to the * as well. Six simultaneous presentations you mentioned here is for VoicePulse, right ? Do you know the limitation for Iconnect ? I can't use VoicePulse because I live in California, and need California numbers. Thanks SW Sorry, I should have been more explicit. Yes, I meant with VoicePulse and as far as I know, with Nufone too. These are the only ones I have checked explicitly (the IAX providers). Also, the number of simultaneous presentations may bary from provider to provider. Vonage may be able to do the same thing as well, but you would have to verify that with them, or someone on the list may also know. As far as I know, iconnect explicitly disallows multiple call presentations. The iconnect thing was discussed on the list a month or two back. Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to detect process 256 frames
Hi folks, Does anybody have any idea what this is; WARNING[1232119104]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 256 frames WARNING[1232119104]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 256 frames WARNING[1232119104]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 256 frames WARNING[1232119104]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 256 frames WARNING[1232119104]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 256 frames WARNING[1232119104]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 256 frames WARNING[1232119104]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 256 frames WARNING[1232119104]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 256 frames WARNING[1232119104]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 256 frames WARNING[1232119104]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 256 frames WARNING[1232119104]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 256 frames I see this all over when I make a call from SIP to H323 (chan_h323) in pass-through mode. Cheers SW ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to detect process 256 frames
Thanks, Jeremy, that was indeed the problem. Message: 2 Date: Thu, 18 Dec 2003 12:56:48 -0500 From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Unable to detect process 256 frames Reply-To: [EMAIL PROTECTED] SW wrote: Hi folks, Does anybody have any idea what this is; WARNING[1232119104]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 256 frames Do not try to do inband DTMF on G.729 Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G729 question
Hi Clif, My experience with G.729 and asterisk is not good. My first registration was good, it worked. Then I bought more license and tried to upgrade it, it blew everything off. Still waiting Digium support to give me a helping hand. If you use pass-through feature then I guess you are fine. I have SIP users going to h.323 g/w and I need g.729. So now I have it in pass-through mode, I think that requires less CPU overhead and I do not have to mess with licenses. Cheers SW Message: 5 Date: Thu, 18 Dec 2003 13:59:06 -0500 From: Clif Jones [EMAIL PROTECTED] To: asterisk users [EMAIL PROTECTED] Subject: [Asterisk-Users] G729 question Reply-To: [EMAIL PROTECTED] I am thinking about using the G729 codecs on my endpoint devices and purchasing some G729 licenses for Asterisk but I have several questions: 1. Which G729 codec is sold by Digium for Asterisk, G729, G729A, B...I? 2. If I have G729A on one end and G729B on the other, are they compatible? I have looked all over the place for question 2, but without buying the ITU docs I cannot seem to find this answer. I know that G729A is low complexity which seems to be what Cisco 7960's use but I have some others that support G729B which has comfort noise and reduced transmission during silence. If anyone knows how the different G729 codecs interoperate I would be eager to know. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: G729 question
Hi Kannaiyan, I did that five days ago, still waiting Digium to provide me with the combined license key. I think Digium has to go to the license owner for this. thanks SW -Original Message- From: Kannaiyan Natesan [mailto:[EMAIL PROTECTED] Sent: Thursday, December 18, 2003 5:35 PM To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: G729 question Hi SW, The procedure is, 1. Submit the old registration key 2. Get the new License Key. ( Old + New ) 3. Then install with a single registration key. (You can try to ask for the procedure how to destroy your previous license, that will make asterisk clean) If you try to upgrade the existing license then it will blew of even the existing licensel. You can contact Digium and submit your previous key and get a new key which can give you the total licenses in a single key. Kannaiyan Hi Clif, My experience with G.729 and asterisk is not good. My first registration was good, it worked. Then I bought more license and tried to upgrade it, it blew everything off. Still waiting Digium support to give me a helping hand. If you use pass-through feature then I guess you are fine. I have SIP users going to h.323 g/w and I need g.729. So now I have it in pass-through mode, I think that requires less CPU overhead and I do not have to mess with licenses. Cheers SW ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] x100P incoming
Hi Gurus How do I make x100P does not answer incoming calls ? I want * play dead for incoming calls. I do not have any context for incoming calls from x100p, in zapata.conf. Call also get logged into the CDR, that too I do not want. I am using x100p for outgoing calls only. Any help appreciate. Cheers SW ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] g729 error - WARNING[1074433504]:
Hi, I just applied four new g729 license to my * installation. Registration was successful == NOW, PLEASE ANSWER THE FOLLOWING QUESTION: Do you accept the terms of this agreement? yes(y) or no(n)y ...Please wait a few seconds Registration successful! == But, Now I cant start *, it comes up with the following error; [codec_g729b.so] = (Annex B (floating point) G.729/PCM16 Codec Translator) WARNING[1082825920]: File asterisk.c, Line 234 (listener): Select retured error: Interrupted system call WARNING[1082825920]: File asterisk.c, Line 234 (listener): Select retured error: Interrupted system call shmget: Invalid argument WARNING[1074433504]: File codec_g729b.c, Line 511 (load_module): Unable to initialize va stuff: -1 == Detected 4 licensed G.729 transcoders WARNING[1074433504]: File translate.c, Line 219 (calc_cost): Translator 'g729tolinb' does not produce sample frames. == Registered translator 'g729tolinb' from format G729A to SLINR, cost 9 Error Opening channel:0 call va_g729_init_global(..) prior to open! WARNING[1074433504]: File codec_g729b.c, Line 202 (lintog729_new): No available g729 resources for channel 0 WARNING[1074433504]: File codec_g729b.c, Line 342 (lintog729_framein): G729 resources are not allocated, exiting Error Opening channel:0 call va_g729_init_global(..) prior to open! WARNING[1074433504]: File codec_g729b.c, Line 202 (lintog729_new): No available g729 resources for channel 0 WARNING[1074433504]: File codec_g729b.c, Line 342 (lintog729_framein): G729 resources are not allocated, exiting Error Opening channel:0 call va_g729_init_global(..) prior to open! WARNING[1074433504]: File codec_g729b.c, Line 342 (lintog729_framein): G729 resources are not allocated, exiting Error Opening channel:0 call va_g729_init_global(..) prior to open! WARNING[1074433504]: File codec_g729b.c, Line 202 (lintog729_new): No available g729 resources for channel 0 Appreciate any help. Cheers SW ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip registration send out by asterisk
Hi friends, I've noticed that first register message sent by * always get rejected by the destination sip server. Then * sends a second registration message ( with Autherization section, and that get accepted by the destination host). Why is this ? Isnt there a way to tell * to send with Autothorization message the first attempt ? Asterisk sends this first 9 headers, 0 lines 11 headers, 0 lines Reliably Transmitting: REGISTER sip:sipauth.deltathree.com SIP/2.0 Via: SIP/2.0/UDP 66.xxx.xxx.xxx:5060;branch=z9hG4bK6b37ba4a From: sip:[EMAIL PROTECTED];tag=as3e96887d To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 106 REGISTER User-Agent: Asterisk PBX Expires: 160 Contact: sip:[EMAIL PROTECTED] Event: registration Content-length: 0 After 401 unautorized from iconnect asterisk sends this 8 headers, 0 lines 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sipauth.deltathree.com SIP/2.0 Via: SIP/2.0/UDP 66.xxx.xxx.xxx:5060;branch=z9hG4bK6b37ba4a From: sip:[EMAIL PROTECTED];tag=as3e96887d To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 107 REGISTER User-Agent: Asterisk PBX Authorization: Digest username=1510xxx, realm=deltathree.com, algorithm=MD5, uri=sip:sipauth.deltathree.com, nonce=3fdecbbf, response=49558c95bc3383bcbf76a26376e1614a Expires: 160 Contact: sip:[EMAIL PROTECTED] Event: registration Content-length: 0 Cheers SW ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iconnect incoming works, so what ...
Hi, I would like to ask anyone, who use Iconnect incoming calls, receiving via a context other than [general] to respond to this message. I challenge that there is no way that we can do with current * code or the way SIP messages are received from iconnect. No matter what you do, iconnect calls always land in the general section. Best source of info out there, John Todds example, sip.conf is also configured to receive iconnect calls through general section. When I post here people would advise (appreciate), but I would like to hear from any one who has their * sip.conf configured so that iconnect calls find their context ? Just to summarize, from Iconnect I get three parameters 1. Username (52671573) 2. Password (9876) 3. phone number (1411212) so in sip.conf, we have the register string; register=1411212:[EMAIL PROTECTED]/1411212 and the context for the peer/friend (a) [iconnect] type=friend secret=9876 username=52671573 host=sipauth.deltathree.com dtmfmode=inband context = iconnect-in (b) [sipauth.deltathree.com] type=friend secret=9876 username=52671573 host=sipauth.deltathree.com dtmfmode=inband context = iconnect-in (a) or (b) above will function as the incoming context for iconnect call. As a side note, method (b) works for incoming calls from fwd. Cheers SW ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] command Authenticate
Hi folks, Sorry to post bunch of messages today, I am deep into this and hope any help from the experts. I am using the command Authenticate as explained in wi-ki: === Authenticate(password[|options]) Requires a user to enter a given password in order to continue execution. If the password begins with the '/' character, it is interpreted as a file which contains a list of valid passwords (1 per line). An optional set of options may be provided by concatenating any of the following letters: a - Set account code to the password that is entered = exact statement in my extensions.conf is exten = s,1,Authenticate(1234|abc) So I get the voice prompt for the password, I send the digits, * authenticate, and call goes through = all are happy. Then I go to the CDR and I find the accountcode is logged as 1234. I would expect it to be abc ? If I do this, exten = s,1,Authenticate(1234|10) nothing get logged? Is this the way it supposed to work, or am I doing something wrong here ? Cheers SW ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voice mail - sip:notify message
Thanks, John. My requirement here is little different. I am using * as a Voice Mail server for Vocal ((in addition It does codec conversion and routing to PSTN/SIP and PSTN/H323). Thing is Vocal doesn't seems to like the Notify message coming from user asterisk. If I can modify this I will have * seamlessly providing full capabilities of voice mail to Vocal . I can't find a way to get the VMWI to pass through to a SIP client of Vocal. Cheers SW = Message: 6 Date: Sun, 14 Dec 2003 13:33:09 -0500 To: [EMAIL PROTECTED] From: John Todd [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] voice mail - sip:notify message Reply-To: [EMAIL PROTECTED] At 12:29 AM -0800 12/13/03, SW wrote: Hi folks, To provide MWI, * will send out a sip:notify message to the UA. The originator of this message is asterisk, as shown below; NOTIFY sip:[EMAIL PROTECTED]:5065 SIP/2.0 Via: SIP/2.0/UDP 66.121.xxx.yyy:5060;branch=z9hG4bK0466cb21 From: asterisk sip:[EMAIL PROTECTED];tag=as0ffb1bdc === To: sip:[EMAIL PROTECTED]:5065 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 38 Messages-Waiting: yes Voicemail: 4/11 Is there a way that I can change this 'originator' to a numeric value ? Cheers SW SW - See the below conversation from April regarding the same topic. I don't think this became a patch or made it into CVS. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voice mail - sip:notify message
Hi folks, To provide MWI, * will send out a sip:notify message to the UA. The originator of this message is asterisk, as shown below; NOTIFY sip:[EMAIL PROTECTED]:5065 SIP/2.0 Via: SIP/2.0/UDP 66.121.xxx.yyy:5060;branch=z9hG4bK0466cb21 From: asterisk sip:[EMAIL PROTECTED];tag=as0ffb1bdc === To: sip:[EMAIL PROTECTED]:5065 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 38 Messages-Waiting: yes Voicemail: 4/11 Is there a way that I can change this 'originator' to a numeric value ? Cheers SW ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] simple question on sip.conf
Hi folks, I want to fix hole in my asterisk set up. I use Vocal as my sip proxy and * for voice mail and the g/w to PSTN, Iconnect, fwd etc. So from Vocal I redirect sip requests which needs to go 'other' places. This senario works fine. Now the issue is someone else running a vocal or another SIP proxy can redirect his calls to my * as well. Those calls two will come through general section of the sip.conf and could land on the PSTN as well. (do not ask my *'s IP address folks, I am not going to run a free PSTN g/w :) So, how do I prevent other than my own proxy to use the general section of the sip.conf file ? As a mater of fact all calls from fwd and iconnect two land on the general section. Hope some one can shead some light. Cheers SW Here is my sip.conf [general] ;calls arrive from sip lands here port=5060 context=default-in disallow=all allow=ulaw allow=alaw allow=g729 maxexpirey=180 defaultexpirey=160 ;Connect to Free World Dialup (no NAT) register=61358:[EMAIL PROTECTED]/61358 ;Connect to iconnect register=15108688610:[EMAIL PROTECTED]/15108688610 canreinvite=no [iconnect] ;incoming does not land here, why ? outgoing is fine type=friend secret= username= host=sipauth.deltathree.com dtmfmode=inband ; required by iconnect context=iconnect-in canreinvite=no allow=alaw allow=ulaw allow=g729 [fwd] ;incoming does not land here, why ? outgoing is fine type=friend secret=xxx username=61358 host=fwd.pulver.com context=fwd-in allow=alaw allow=ulaw [vocal] ;used when dialed in from vocal not working type=friend host=ip of vocal server disallow=all allow=g729 allow=ulaw allow=alaw port=5060 canreinvite=no context=vocal-in [vocal-out] ;used to dial out to vocal type=friend host=ip of vocal server allow=g729 allow=ulaw allow=alaw port=5065 canreinvite=no [6300] type=friend username=6300 context=intern ;secret=blah host=dynamic ;defaultip=192.168.254.4 dtmfmode=info nat=1 [6301] type=friend username=6301 host=dynamic dtmfmode=inband context=intern nat=1 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RH9 and h323.conf
Tjapko,
I am kinda new here, too. Let me try to help you out here.
First, most people would not give a second look at your posting because
there are tons of past reference in the list advising people not to use
openh323 1.12.0, pwlib 1.5.0. If you do that you are on your own. That's
what README says.
Also there were many postings advising not to use redhat9. So best bet is to
follow exactly what is said in the Readme file.
Having said that, I did install chan_h323 on a redhat 9, openh323 1.12.0 and
pwlib 1.5.0. And it worked :). What I am doing is allowing my SIP clients to
dial out through h.323. So I do not know other stuff would work or not (like
gatekeeper registration, incoming towards * etc.).
Dial string that you are supposed to use in extensions.conf to call out to a
g/w is as follows
exten = 91x,1,Dial(h323/${Exten:[EMAIL PROTECTED])
This did not work for me, so I tried
exten = 91x,1,Dial(h323/h323:${Exten:[EMAIL PROTECTED]) and it worked.
I do not know what the best cause of action for you, I am just sharing my
experience.
Hope this will help you.
Cheers
SW
Message: 7
Date: Fri, 12 Dec 2003 17:11:18 -0400
From: its Consultancy [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] RH9 and h323.conf
Reply-To: [EMAIL PROTECTED]
Hello everybody,
First time installer and I need the lists advice. My plan is to use
asterisk PBX with some hardware to terminate my calls coming from
several operational gnugk gatekeepers.
Do have RH9 and downloaded the latest asterisk from CVS. Compiled
according instructions and is running fine.
Could hardly find any info on h323 implementation untill the REAME in
the channels directory.
I am an experienced openh323 gatekeeper user so do have good compiled
libraries of openh323 and pwlib on my box. Version 1.5.0 and 1.12.0 to
be exact.
Ran the make in the h323 directory and compile seems to run fine.
Than I ran the make install in the asterisk library again and this also
went without problems.
Ran asterisk but problems finding a h323.conf file. Since there is NO
sign of a h323.conf file in the etc/asterisk directory I copied this one
manually from the channels/h323 directory.
After starting asterisk again with a unconfigured default h323.conf ini
file it seems to run ok ; the message is this :
[chan_h323.so] = (The NuFone Network's Open H.323 Channel Driver)
== Parsing '/etc/asterisk/h323.conf': Found
== Creating H.323 Endpoint
== Registered channel type 'H323' (The NuFone Network's Open H.323
Channel Driver)
== H.323 listener started
== Parsing '/etc/asterisk/enum.conf': Found
Asterisk Ready.
Before I go further configuring the PBX and configuring the gatekeeper
that has to route the calls to the PBX I would like to know if there is
anybody with experience on RH9 that has good results with this (h323)
configuration and if my whole way of implementing the thing was OK?
Furthermore I would like to know if there is a good h323 implementation
guide somewhere available like an extension on the Asterisk Handbook
Project that has some very usefull information but lacking h323
information. ( I read the 71 pages!)
Thanks, Tjapko.
___
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[Asterisk-Users] unknown RTP codec 19
Hi Folks, I had success with making SIP to H323 g/w calls using chan_h323 and G.729. Calls go through well though it takes incredibly long time to get connected which I thing due to fast start being disabled at the remote end, will ask remote side to fix it. There is a Notice message which appears on my * console which bothers me. While in a conversation, I get the following; File rtp.c line 418 (ast_rtp_read): Unknown rtp codec 19 received. I read in the mail archive that rtp codec 19 is comfort noise. Should I ignore this notice ? Or is there a better way to tell * asterisk to handle comfort Noise ? Cheers SW ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 readme file
Hi Brian, I will first use the listed versions as you suggested. It is great if the readme file is corrected. Instead of You first need the latest release versions of PWLib and Open H.323 from ..., it should read You first need the LISTED release versions of PWLib and Open H.323 from Then poor newbies do not need to get RTFM'ed :). Also please take a look at this posting; http://lists.digium.com/pipermail/asterisk-cvs/2003-November/000498.html So there were changes in mid Novemeber time frame. Most notably the referance to G729 has been taken away. A big requirement to run Open h323 v1.11.7 was that the it was the last version with G729. So chan_h323 must have been modified that it can now use G729 (if license is bought from Digium) without depending on open h323. Please correct me if I am wrong. Cheers SW Date: Mon, 8 Dec 2003 22:59:05 -0600 (CST) From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Digium. Com [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] chan_h323 readme file Reply-To: [EMAIL PROTECTED] If you have problems use the latest ones. Otherwise use whats listed because thats what JJ has tested and is known to work. bkw On Mon, 8 Dec 2003, SW wrote: Hello I am getting ready to install chan_h323. Just updated my * with the latest code from CVS (12/08/03). I was reading the Readme file and confused. Quoted from the README NOTICE: Whatever you do, DO NOT USE distrubution specific installs of Open H.323 and PWLib. In fact you should check to make sure your distro didn't install them for you without your knowledge. Check everything out of CVS. If you dont know how to deal with cvs, learn. Also, if you are not using the listed versions of Open H.323 or PWlib you are on your own, sorry. To compile this code: You first need the latest release versions of PWLib and Open H.323 from http://www.openh323.org/. Make sure you follow the build instructions EXCPLICTLY. Unquote First para says that I should only use the listed versions of PWlib and open h.323 (v1.4.11 and v1.11.7) Then it says that I should use the latest release versions of PWLib and Open H.323 (v1.5.2 and v1.12.2.) I've read Wi-Ki and many other places that Chan_H323 use open h323 v1.11.7, but wondering whether JJ has made some changes lately :) What should be the correct way ??? Thanks SW ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323 readme file
Hello I am getting ready to install chan_h323. Just updated my * with the latest code from CVS (12/08/03). I was reading the Readme file and confused. Quoted from the README NOTICE: Whatever you do, DO NOT USE distrubution specific installs of Open H.323 and PWLib. In fact you should check to make sure your distro didn't install them for you without your knowledge. Check everything out of CVS. If you dont know how to deal with cvs, learn. Also, if you are not using the listed versions of Open H.323 or PWlib you are on your own, sorry. To compile this code: You first need the latest release versions of PWLib and Open H.323 from http://www.openh323.org/. Make sure you follow the build instructions EXCPLICTLY. Unquote First para says that I should only use the listed versions of PWlib and open h.323 (v1.4.11 and v1.11.7) Then it says that I should use the latest release versions of PWLib and Open H.323 (v1.5.2 and v1.12.2.) I've read Wi-Ki and many other places that Chan_H323 use open h323 v1.11.7, but wondering whether JJ has made some changes lately :) What should be the correct way ??? Thanks SW ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Deltathree icomming problem
Hi Chris, This is how my iconnect incomming is setup. register = 12047440600:[EMAIL PROTECTED]/12047440600 in internal context I have exten = 12047440600,1,Dial(SIP/1000) Try simplifying your config first. Sathya From: Chris HARIGA [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Deltathree icomming problem Date: Sat, 29 Nov 2003 10:15:54 -0500 Reply-To: [EMAIL PROTECTED] I'n not behind a NAT. I know the URL http://www.loligo.com/asterisk and I setup like there and is the same :(( I use sipauth.deltathree.com witch is the IP from my sip.conf :) Best regards, Chris HARIGA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Resend: Help for oh323
Michael,
Thanks a bunch, I downloaded from inaccessnetworks.com thinking that it is
the latest :). Ok I will upgrade it. just for the record, following worked.
exten = _87.,1,Dial(OH323/H323:${EXTEN:[EMAIL PROTECTED])
Cheers
Sathya
Date: Fri, 28 Nov 2003 11:28:59 +0200
From: Michael Manousos [EMAIL PROTECTED]
Organization: inAccess Networks
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Resend: Help for oh323
Reply-To: [EMAIL PROTECTED]
Hi Sathya,
I bet you use OpenH323 v1.12.0.
Go to v1.12.2 and you will be OK.
There isn't anything wrong with your syntax, it's an
OpenH323 issue.
Michael.
SW wrote:
anyone who can shed some light ? Or oh323 is completely dumped and I
should
go to chan_h323 ?
-Original Message-
From: SW [mailto:[EMAIL PROTECTED]
Sent: Thursday, November 27, 2003 8:28 AM
To: [EMAIL PROTECTED] Digium. Com
Subject: Help for oh323
Hi Friends,
Hope you would help me out here, I have searched the asterisk
user list for hours and also read the readme and test files that
comes with the driver. I need a very simple scenario. I have SIP
clients and want to use oh323 to dial out to PSTN using a h323 gateway.
a)If I set the extention.conf like this:
exten = _87.,1,Dial(OH323/16.52.153.206)
oh323 dials out (I can ring a netmeeting client at 16.52.153.206).
(b)But if I set it like this, oh323 will not dials out ?
exten = _87.,1,Dial(OH323/${EXTEN:[EMAIL PROTECTED])
In summary what I am trying to achieve is the following;
Lets say Sip user dial 915105418168, then I want 9 to be dropped
and the extension information to be send to the g/w at
16.52.153.206. Isn't exten =
_9x,1,Dial(OH323/${EXTEN:[EMAIL PROTECTED]) is the right
way ?. Why is this not working ?
I must be doing a wrong syntaxt, but couldnt find where I go wrong.
I am attaching the trace for above two cases, please help ?
Cheers
Sathya
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[Asterisk-Users] Help for oh323
Hi Friends,
Hope you would help me out here, I have searched the asterisk
user list for hours and also read the readme and test files that
comes with the driver. I need a very simple scenario. I have SIP
clients and want to use oh323 to dial out to PSTN using a h323 gateway.
a)If I set the extention.conf like this:
exten = _87.,1,Dial(OH323/16.52.153.206)
oh323 dials out (I can ring a netmeeting client at 16.52.153.206).
(b)But if I set it like this, oh323 will not dials out ?
exten = _87.,1,Dial(OH323/${EXTEN:[EMAIL PROTECTED])
In summary what I am trying to achieve is the following;
Lets say Sip user dial 915105418168, then I want 9 to be dropped
and the extension information to be send to the g/w at
16.52.153.206. Isn't exten =
_9x,1,Dial(OH323/${EXTEN:[EMAIL PROTECTED]) is the right
way ?. Why is this not working ?
I must be doing a wrong syntaxt, but couldnt find where I go wrong.
I am attaching the trace for above two cases, please help ?
Cheers
Sathya
Traces for both cases are given below;
0:00.076 OpenH323 Wrapper OpenH323 WrapperVersion
0.0alpha
0 by inAccess Networks (www.inaccessnetworks.com) on Unix Linux
(2.4.20-8-i686)
at 2003/11/26 20:15:07.622
0:00.078 OpenH323 Wrapper H323Created endpoint.
0:00.078 H323 Cleaner H323Started cleaner thread
0:00.212 OpenH323 Wrapper H323Started listener
Listener[ip$*:1720]
0:00.214H323 Listener:8115c18 H323Awaiting TCP connections on
port 1720
0:00.214 OpenH323 Wrapper H323UDP Binding to interface:
0.0.0.0:1
0:00.243 OpenH323 Wrapper H323Added capability:
G.711-ALaw-64k{hw} 1
0:00.279 OpenH323 Wrapper H323Added capability:
UserInput/hookflash 2
0:00.279 OpenH323 Wrapper H323Added capability:
UserInput/basicString 3
0:00.279 OpenH323 Wrapper H323Added capability:
UserInput/dtmf 4
0:00.279 OpenH323 Wrapper H323Added capability:
UserInput/RFC2833 5
4:00.687 ThreadID=0x4a774440 H323Making call to:
[EMAIL PROTECTED]
4:00.688 ThreadID=0x4a774440 H323Attempt to use invalid URL
[EMAIL PROTECTED]
4:00.688 ThreadID=0x4a774440 H323Could not parse
[EMAIL PROTECTED]
4:00.757 ClearCallT...d:0812ab10 H323Attempt to clear unknown
call
8:14.840 ThreadID=0x4a774440 H323Making call to:
16.52.153.206
8:14.904 ThreadID=0x4a774440 H323Added capability:
G.711-ALaw-64k{hw} 1
8:14.905 ThreadID=0x4a774440 H323Added capability:
UserInput/hookflash 2
8:14.905 ThreadID=0x4a774440 H323Added capability:
UserInput/basicString 3
8:14.905 ThreadID=0x4a774440 H323Added capability:
UserInput/dtmf 4
8:14.905 ThreadID=0x4a774440 H323Added capability:
UserInput/RFC2833 5
8:14.905 ThreadID=0x4a774440 H323Found capability:
G.711-ALaw-64k{hw} 1
8:14.905 ThreadID=0x4a774440 H323Found capability:
UserInput/hookflash 2
8:14.905 ThreadID=0x4a774440 H323Found capability:
UserInput/basicString 3
8:14.906 ThreadID=0x4a774440 H323Found capability:
UserInput/dtmf 4
8:14.906 ThreadID=0x4a774440 H323Found capability:
UserInput/RFC2833 5
8:14.906 ThreadID=0x4a774440 RFC2833 Handler created
8:14.906 ThreadID=0x4a774440 H323Added capability:
G.711-ALaw-64k{hw} 1
8:14.907 ThreadID=0x4a774440 H323Created new connection:
ip$localhost/25259
8:14.908 H225 Caller:80f4688 H225Started call thread
8:15.064 H225 Caller:80f4688 H323TCP Could not connect to
16.52.153.2
06:1720 (local port=1) - No route to host(113)
8:15.065 H225 Caller:80f4688 H323Clearing connection
ip$localhost
/25259 reason=EndedByConnectFail
8:15.065 H225 Caller:80f4688 H323Call end reason for
ip$localhost
/25259 set to EndedByConnectFail
8:15.066 H225 Caller:80f4688 H225Sending release complete
PDU: ca
llRef=25259
8:15.200 H225 Caller:80f4688 H323Clearing connection
ip$localhost
/25259 reason=EndedByTransportFail
8:15.200 H323 Cleaner H323Cleaning up connections
8:15.201 H323 Cleaner H323Connection
ip$localhost/25259 cl
osing: connectionState=NoConnectionActive
8:15.201 H323 Cleaner H323H323Transport::Close
8:15.201 H323 Cleaner H323
H323Transport::CleanUpOnTerminat
ion for H225 Caller:80f4688
8:15.201 H323 Cleaner H323Connection
ip$localhost/25259 te
rminated.
8:15.203 H323 Cleaner H323Connection
ip$localhost/25259 de
leted.
8:15.204 H323 Cleaner H323Cleaning up connections
8:15.369 ClearCallT...d:080f1a48 H323Attempt to clear unknown
call ip
$localhost/25259
___
Asterisk-Users
[Asterisk-Users] g729 license
Hello, I am trying to see what I need to do SIP to H323 using G.729. I have Oh323 and SIP working with G711 fine. If I have a SIP client configured to use G729 and H323 client also to G729, how many license should I need to buy from Digium. Many thanks SW ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
