[Asterisk-Users] Incorrect CDRs
Hello list, We are having some serious problems with CDR and billing. CDR shows that some of the (unanswered) calls were lasted for 2-3 days. This is the situation: We have 2 Asterisk servers, connected to PSTN. Phone A -- Asterisk1 -- Gateway - PSTN Phone B -- Asterisk2 -- Gateway - PSTN A dials B, call reaches B, but it rejects the call, and sends back BUSY. The CDR on B is perfect, it says BUSY. Following is from the CDR file of Asterisk2: start ,answer ,end,dur ,billable,disposiion,ama flags,, - 2/8/2005 1:06,2/8/2005 1:06, ,10,0 ,BUSY ,DOCUMENTATION But the corrosponding record in Asterisk1 says something different: start,answer ,end ,dur ,billable,disposiion,ama flags,, - 2/8/2005 1:06,2/8/2005 1:06,2/11/2005 1:51,261895,261882 ,ANSWERED ,DOCUMENTATION As you can see the billable seconds (as per the CDR) are 261882 and the call lasted for 3 days. As you can understand, this is a very serious problem. Can anyone please clarify the following? 1. Why the first Asterisk did not recognise the busy sent back from Asterisk2? 2. Has anyone faced such problems? If so how did they resolve this? 3. What precautions need to be taken to avoid this in the future? I greately appreciate any answer you can provide on this. TIA, Girish __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lost admin password on Polycom IP500?
Hi, --- Matt Darnell [EMAIL PROTECTED] wrote: Does anyone know how to default the admin password on a Polycom IP500? Can you try upgrading/downgrading the bootrom version of the phone? I think that will set the phone settings to factory defaults. After that you can switch back to the bootrom version you are using. Regards, Girish __ Do you Yahoo!? Send a seasonal email greeting and help others. Do good. http://celebrity.mail.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SJPhone SIP Tab
Hi, --- Norman Zhang [EMAIL PROTECTED] wrote: I'm following, http://www.voip-info.org/wiki-Asterisk+phone+sjphone. However, I cannot find the SIP tab. Would someone please give me a few pointers? The screen capture can be seen at URL below The wiki talks about an older version of SJPhone. The one you are using is the latest version (with stun support). Click on the profiles tab and create a new profile. You'll get an options dialog in which you can set the values. Regards, Girish __ Do you Yahoo!? Yahoo! Mail - Easier than ever with enhanced search. Learn more. http://info.mail.yahoo.com/mail_250 ___ 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] software phones for Asterisk - is there a list?
Hello, --- Tomasz Chmielewski [EMAIL PROTECTED] wrote: Is there a list of software phones which will work with Asterisk? See the 'SIP Phones (SIP User Agents)' section here: http://pernau.at/kd/voip/bookmarks-sip-rtp-ua.html Regards, Girish __ Do you Yahoo!? The all-new My Yahoo! - What will yours do? http://my.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] Illegal Instruction (Solved)
Hi all, I solved the 'Illegal Instruction' problem. This is what i did, hope this might help someone later... From my /proc/cpuinfo file: model name : VIA Samuel 2 I found this entry in the Asterisk Makefile and uncommented it: # Pentium VIA processors optimize # PROC=i586 Recompiled.. and now everything is OK. -Girish (Happy) --- Girish Gopinath [EMAIL PROTECTED] wrote: Hi Matt, --- Matt Gibson [EMAIL PROTECTED] wrote: did you delete your old asterisk's modules directory and try again? Thanks for the response. Yes, I removed everything before installing. Including /usr/lib/asterisk/modules and all directories under /var/lib/asterisk. But no luck. -Girish Sorry for the cross-post. I posted this to the -users list about 12 hours back and havent got any reply. Probably nobody there had experienced this problem. Can someone take a look into this and tell me why Asterisk seg-faults? --- Girish Gopinath [EMAIL PROTECTED] wrote: Folks, I have an RH machine which was running Asterisk 1.0-RC1. This evening i switched to Asterisk 1.0.1. Installation was successful, however Asterisk terminates abnormally during startup flashing an 'Illegal Instruction' message on the console. I noticed that this happens while loading the iax2 module. I am attaching the trace of core with this. Can anyone tell me what is going wrong and how to fix it? TIA, Girish From the console: [EMAIL PROTECTED] asterisk-1.0.1]# asterisk -cvvv == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk 1.0.1, Copyright (C) 1999-2004 Digium. . . [chan_iax2.so] = (Inter Asterisk eXchange (Ver 2)) == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) == Using TOS bits 16 == IAX Ready and Listening on 0.0.0.0 port 4569 Illegal instruction (core dumped) __ Do you Yahoo!? Check out the new Yahoo! Front Page. www.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP REGISTER -- Asterisk non-compliant or is it the provider?
Hello, --- Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] wrote: The provider's support staff says that the userid in 'From: sip:[EMAIL PROTECTED] ...' should be the phone number while the userid in 'Authorization: Digest username=userid...' of the same REGISTER message should be the account name. I am not sure if this can be right. At least, whether compliant or not, it would seem that such a REGISTER message cannot be constructed by Asterisk. The From header field in a SIP message is used to identify the initiator of the request. AFAIK, the 050 in the From header acts as a display name. It can be used to determine the processing rules by other SIP entities. For example, PSTN gateways can use it to determine if it is a valid callerid or not (For INVITEs). The Auth credentials in the Request can be different. It should be the username and password for the account that the provider has given you. I hope others here will give a better explanation on this... - is it in compliance with RFC3261 to have different values in the From and the Digest username fields? I think yes. Our UACs register with SIP Express Router with different values in these fields. Attached below is an ngrep trace of such a request processing. - can Asterisk construct such a REGISTER message? Sorry. I am not sure about this. Regards, Girish Here's the trace: REGISTER sip:XXX.XXX.XXX.XXX SIP/2.0. Via: SIP/2.0/UDP 192.168.68.24:12894. Max-Forwards: 70. From: sip:[EMAIL PROTECTED];tag=699fdcaedde144a68097a86c5ec00655;epid=7de641515e. To: sip:[EMAIL PROTECTED]. Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER. Contact: sip:192.168.68.24:12894;methods=INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER. User-Agent: RTC/1.2.4949 Event: registration. Allow-Events: presence. Content-Length: 0. . SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.68.24:12894;rport=4061;received=XX.XX.XX.XXX. From: sip:[EMAIL PROTECTED];tag=699fdcaedde144a68097a86c5ec00655;epid=7de641515e. To: sip:[EMAIL PROTECTED];tag=b27e1a1d33761e85846fc98f5f3a7e58.d1fd. Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER. WWW-Authenticate: Digest realm=XXX.XXX.XXX.XXX, nonce=418cd4b94a2ea94004191fd618b7bbb7041f8f40, qop=auth. Server: SIP EXpress Router (0.8.14 (i386/linux)). Content-Length: 0. . . REGISTER sip:XXX.XXX.XXX.XXX SIP/2.0. Via: SIP/2.0/UDP 192.168.68.24:12894. Max-Forwards: 70. From: sip:[EMAIL PROTECTED];tag=699fdcaedde144a68097a86c5ec00655;epid=7de641515e. To: sip:[EMAIL PROTECTED]. Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER. Contact: sip:192.168.68.24:12894;methods=INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER. User-Agent: RTC/1.2.4949 Authorization: Digest username=girish-smarttest-com, realm=XXX.XXX.XXX.XXX, qop=auth, algorithm=md5, uri=sip:XXX.XXX.XXX.XXX, nonce=418cd4b94a2ea94004191fd618b7bbb7041f8f40, nc=0001, cnonce=15420645451543562242791578613325, response=89d8531e598629b022230df475b5bb65. Event: registration. Allow-Events: presence. Content-Length: 0. . . SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.68.24:12894;rport=4061;received=XX.XX.XX.XXX. From: sip:[EMAIL PROTECTED];tag=699fdcaedde144a68097a86c5ec00655;epid=7de641515e. To: sip:[EMAIL PROTECTED];tag=b27e1a1d33761e85846fc98f5f3a7e58.d1fd. Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER. Expires: 120. Contact: sip:XX.XX.XX.XXX:4061;q=0.00;expires=3600. Server: SIP EXpress Router (0.8.14 (i386/linux)). Content-Length: 0. __ Do you Yahoo!? Check out the new Yahoo! Front Page. www.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Illegal Instruction
Folks,
I have an RH machine which was running Asterisk 1.0-RC1. This evening i switched to
Asterisk 1.0.1. Installation was successful, however Asterisk terminates abnormally
during startup flashing an 'Illegal Instruction' message on the console. I noticed that
this happens while loading the iax2 module. I am attaching the trace of core with this.
Can anyone tell me what is going wrong and how to fix it?
TIA, Girish
From the console:
[EMAIL PROTECTED] asterisk-1.0.1]# asterisk -cvvv
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.0.1, Copyright (C) 1999-2004 Digium.
.
.
[chan_iax2.so] = (Inter Asterisk eXchange (Ver 2))
== Manager registered action IAXpeers
== Parsing '/etc/asterisk/iax.conf': Found
== Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2))
== Using TOS bits 16
== IAX Ready and Listening on 0.0.0.0 port 4569
Illegal instruction (core dumped)
Here's the trace:
(gdb) bt full
#0 0x402f2e1c in try_firmware (s=0xbfffdfd0 /var/lib/asterisk/firmware/iax/iaxy.bin)
at chan_iax2.c:953
stbuf = {st_dev = 770, __pad1 = 0, st_ino = 1073945564, st_mode = 135379256,
st_nlink = 135379280,
st_uid = 3221217128, st_gid = 1073945564, st_rdev = 581449477212191056, __pad2 =
57192,
st_size = 1073905796,
st_blksize = 135379296, st_blocks = 0, st_atim = {tv_sec = 8, tv_nsec = 1073905676},
st_mtim = {tv_sec = 1075596704,
tv_nsec = 135379336}, st_ctim = {tv_sec = -1073750120, tv_nsec = 1075596704},
__unused4 = 135379256,
__unused5 = 135379347}
cur = (struct iax_firmware *) 0xbfffdfef
ifd = -1
fd = 25
res = -1073750033
fwh = (struct ast_iax2_firmware_header *) 0x0
fwh2 = {magic = 1818914, version = 0, devname =
\000\000x???[EMAIL PROTECTED], datalen = 854033,
chksum = [EMAIL PROTECTED], data = 0xbfffdf1a }
md5 = {buf = {3221217232, 3221217271, 3221217487, 3221217232}, bits =
{3221217487, 0},
in = '\0' repeats 16 times,
\000\000\000\0008?\021\b\000\000\000\000\000\000\000\000?\001\000\000\r\000\000\000\002\000\000\000\001\000\000\000?\006\000\\020\000}
sum = \001\200??
buf = [EMAIL PROTECTED];[EMAIL PROTECTED]@6k\r\b, '\0'
repeats 12 times, ([EMAIL PROTECTED],
'\0' repeats 12 times,
\;[EMAIL PROTECTED]@|[EMAIL PROTECTED]@\000\000\000\000\000\000\000\0008???\022?\002@
[EMAIL PROTECTED](\033s \000\000\000\000, '\0' repeats 12 times,
?X\033@, '\0' repeats 18 times, u \000\000\000\000\000\000\000\000?X\033@, '\0'
repeats 18 times...
len = -1073750232
chunk = -1073750064
s2 = 0xbfffd9d0
last = 0xbfffdfef iaxy.bin
#1 0x402ee091 in reload_firmware () at chan_iax2.c:1156
cur = (struct iax_firmware *) 0x811b938
curl = (struct iax_firmware *) 0x811b993
curp = (struct iax_firmware *) 0x
fwd = (DIR *) 0x811b938
de = (struct dirent *) 0xbfffdfee
dir =
/var/lib/asterisk/firmware/[EMAIL PROTECTED]@[EMAIL PROTECTED]@[EMAIL
PROTECTED]@[EMAIL PROTECTED]@?1\034@@?\016\b\b?\r\b\033[1;30;40m
== \033[0;37;[EMAIL PROTECTED]@\020\000\000\000 [EMAIL PROTECTED]@3\000\000\---Type
return to continue, or q return to quit---
000s?\016\bj?\002@@\227\016\b$?0@...
fn =
/var/lib/asterisk/firmware/iax/[EMAIL PROTECTED]@[EMAIL PROTECTED]@[EMAIL
PROTECTED]@[EMAIL PROTECTED]@[EMAIL PROTECTED]@[EMAIL PROTECTED]@[EMAIL
PROTECTED]@[EMAIL PROTECTED]@[EMAIL PROTECTED]@[EMAIL PROTECTED]@[EMAIL PROTECTED]...
#2 0x402ed502 in load_module () at chan_iax2.c:7489
config = 0xbfffdfee /iaxy.bin
res = 0
x = 0
iabuf = 0.0.0.0\000x???G\200\002@
reg = (struct iax2_registry *) 0x0
peer = (struct iax2_peer *) 0x0
sin = {sin_family = 2, sin_port = 55569, sin_addr = {s_addr = 0}, sin_zero =
x\033\003@}
#3 0x08057608 in ast_load_resource (resource_name=0x8107bb3 chan_iax2.so) at
loader.c:333
fn = /usr/lib/asterisk/modules/chan_iax2.so\000\000.so\000a.so, '\0' repeats
207 times
errors = 0
res = -1073750034
m = (struct module *) 0x8118d98
flags = 0
val = 0xbfffdfee /iaxy.bin
key = 0x0
o = 135368088
cfg = (struct ast_config *) 0x0
tmp = \033[33;40mInter Asterisk eXchange (Ver
2)\033[0;37;[EMAIL PROTECTED]@\000\000\000
#4 0x08057e1e in load_modules () at loader.c:428
mods = (DIR *) 0x81079b0
d = (struct dirent *) 0x0
x = 1
cfg = (struct ast_config *) 0x8106130
v = (struct ast_variable *) 0x0
tmp =
\033[1;37;40mchan_iax2.so\033[0;37;40m\000\00040m\000;40m\000\000?\004\000\000#1063;\016\b\034\000\000\0008???\220\237\a\b#1063;\016\b??\a\b?T\r\b\2009\r\b
#5 0x0809b965 in main (argc=-1073748336, argv=0xbfffe580) at asterisk.c:1844
---Type return to continue, or q return to quit---
pw = (struct passwd *) 0xbfffe690
c = -1073750034
filename =
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
Hello,
From: Sunrise Ltd [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
Date: Tue, 13 Jul 2004 16:31:58 +0900 (JST)
snip
If Asterisk is directed to speak SIP on port 5061 and SER
remains on port 5060, then how do you get Asterisk to talk
to SER and vice versa?
Would you care to share this with us?
It is something like this:
Asterisk extensions.conf:
[globals]
SERADDRESS=XXX.XXX.XXX.XXX:5060
[context]
exten = yourexten,1,Dial(SIP/[EMAIL PROTECTED],20,r)
In ser.cfg:
if (method == INVITE) {
if (uri =~ sip:[EMAIL PROTECTED]){
log(1, Forwarding to Asterisk\n);
rewritehostportt(XXX.XXX.XXX.XXX:5061);
t_relay();
break;
}
}
rgds
benjk
Regards, Girish
_
Earn without investing. http://go.msnserver.com/IN/52048.asp Sell anything
under the sun.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
Excellent Post! Very Informative. Thanks a lot Sir! Regards, Girish From: Olle E. Johansson [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous Date: Mon, 12 Jul 2004 10:52:33 +0200 Paul Mahler wrote: Well, this is certainly getting exciting. Yes, it is. Sorry for coming in late to this debate... Andy, I took your advice and re-read the RFP. It's actually RFC, not RFP. (teasing :-) So, gentlemen, help me out here. The spec says: The Address of record is the SIP address that the registry knows the registrand. . . The Address of record is the public SIP uri you want people to call you at, regardless of the address of the phone you are answering on. It's the SIP phone address you place on your business card. A client uses the REGISTER method to register the address listed in the TO header field with a SIP Server. A client registers a temporary address, the address to a SIP UA, to the SIP registrar that is responsible for the domain in the AOR. This tells the SIP registrar (or location server) where to find you if someone calls your URI. When sending mail, I am not addressing the mail to the IP address you are reading the mail on, I am using your public e-mail address that is mapped to an e-mail server that is responsible for all e-mail to your domain. Later on, you fetch the e-mail from an e-mail client somewhere, with an IP address that propably changes as you travel around signing books ;-) SIP works the same way. You have a public address and a SIP proxy being responsible for keeping track of where you want to answer your calls. You can surely register several phones that you want to answer on. The proxy takes care of hiding this to the callee, so that the caller only get one set of responses. That's what the forking stuff is all about. If one phone is busy and the other one is answering, we should only signal 200 OK in SIP lingo to the caller. I don't see how two different clients can register with a server as the same address of record. Doesn't the second registration from a new client change the address of record for the registered client? You have one address-of-record that maps into several SIP URIs, one for each device. These are not as long-term as your a-o-r SIP URI. From the RFC: Location Service: A location service is used by a SIP redirect or proxy server to obtain information about a callees possible location(s). It contains a list of bindings of address-of-record keys to zero or more contact addresses. The bindings can be created and removed in many ways; this specification defines a REGISTER method that updates the bindings. If the second client is trying the same registration as the first client, and it's the responsibility of the client to provide the complete list of bindings, how does the second client know the list of bindings for the first client that bound the registration? It's *not* the responsibility of the *client* to provide a list, it's the server that responds with a list, telling the client by the way, these devices are also registred for the same a-o-r. So isn't this the problem * has? The first client registers as the address of record, then the second client comes in with the same registration and becomes the address of record? The address of record does not change because of a registration. The stored address (the contact: header) of where we can reach you (location) changes. And yes, if you have multiple devices registering for the same Asterisk sip [peer] account, it will be changing for each registration. This is not the behaviour of most SIP Proxys. Asterisk is *not* a SIP proxy. It's a SIP registrar and location server. It's a very clever SIP UA. It wants to be in the middle of the call and wants to be in control of each device. This device-slave view doesn't match the SIP architecture. Due to Asterisk's multi-protocol architecture we have to make some compromises in the SIP channel to be able to have some kind of generic view of calls and phones in the core. A SIP proxy is never the end point of a call, it should never handle the media stream. The power is in the edge, in the phones. This is why transfers and other PBX functions is a bit messy with SIP and Asterisk, we are trying to find a way to do it centralized as Asterisk but de- centralized as SIP... I've spent a considerable amount of time investigating support for multiple registrations on one Asterisk sip [peer] account and after learning about Asterisk's architecture come to the conclusion that it is not an easy or even a desirable feature to implement. The architecture of Asterisk is a PBX, and the dial plan and a lot of apps wants to be in control of the device. It may be possible, but will probably lead to a lot of changes to Asterisk, both core and applications, that no other channel will benefit from. A quick hack to support it may lead to a lot of confusion on how to handle other apps. And it's a lot more work than the bounty will cover. I
RE: [Asterisk-Users] Using asterisk as voicemail system for SER
Hello,
Try something like this:
;sip.conf
[5554321]
type=friend
host=dynamic
context=default
mailbox=5554321
;default context of extensions.conf
exten = 5554321,1,Voicemail2(u${EXTEN})
exten = 5554321,2,Hangup
;voicemail.conf
5554321 = 5554321, Gary, [EMAIL PROTECTED]
Regards, Girish
From: gc [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Using asterisk as voicemail system for SER
Date: Wed, 9 Jun 2004 14:59:58 -0400
I ma new to Asterisk.
I'd like to setup * as voicemail system for SER.
Let's say I have an phone number registered in ser as 5554321. When
somebody dial to ser for this number and nobody answer, the ser will
forward the call to asterisk and get into voicemail box 5554321. I already
have asterisk up and running with mysql setup for asterisk voicemail.
Can somebody show me how to do it? Or show me some examples of sip.conf,
voicemail.conf and extensions.conf.
Gary
_
Holiday? Wedding? Refurnishing? http://go.msnserver.com/IN/49356.asp
Fulfill all your dreams with Citibank Ready Cash!
___
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] Incoming call voice data
Hello, You can use the Record application for that. See the Wiki for more details. Not sure of how to restrict the recording to 30 seconds. See here: http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Record Regards, Girish From: Mag [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Incoming call voice data Date: Sun, 06 Jun 2004 10:47:14 -0400 I am fairly new to asterisk. I have my DID set up with voicepulse.com. And when I call my DID I get an asterisk greeting. If I want to program my dialplan, as: IF I press 5, record a 30 second message and save the file as message.foo How can I do that? Thanks _ Marriage? http://www.bharatmatrimony.com/cgi-bin/bmclicks1.cgi?74 Join BharatMatrimony.com for free. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk SER (www.IPTel.org)
Hello, We use SER as SIP proxy/registrar and Asterisk as the media Server for handling IVRs and Voicemails. The system works fine and the benefit is it is highly scalable. Regards, Girish From: usedcanon [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk SER (www.IPTel.org) Date: Thu, 3 Jun 2004 23:24:07 +0100 What would be the main benefit of this combination ? Do you expect SER to handle more registration traffic etc ? Thanks Umar. _ Attention Hotmail users! Job, love, marriage prospects; http://server1.msn.co.in/sp04/astroyogi/ Get it all in your mailbox! ___ 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/IAX Clients for Linux
Hi, Here's a sip sphone from zultys.com, LIPZ 4. I have tested that, and had no issues. But available only in binary form: http://www.zultys.com/lipz4_download.asp Regards, Girish From: Karl Dyson [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Sip/IAX Clients for Linux Date: Tue, 25 May 2004 10:31:35 +0100 I hear sjphone from sjlabs.com is usable... -Original Message- From: [EMAIL PROTECTED] Hi There, i think all VOIP clients for Linux are unusable! i got testet: Linphone + Linphonec all in version 12.2 Kphone gophone and other... the only programm that is usable is gnomemeeting... does anybody knew some other tools? Best Regards, Mark _ Best of Indian handicrafts. http://www.fabmall.com/affiliatehtml/redir/nl8.asp At MSN Shopping. ___ 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] Re: Asterisk Proxy Type
Asterisk is not a SIP Proxy, It's a soft PBX. But it is a SIP registrar, and forwarding is stateful, i think. I could be wrong. Regards, Girish From: nicolas [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Asterisk Proxy Type Date: Mon, 17 May 2004 11:31:16 +0200 may not correct but i tought * is not a proxy. Ignace CARIA wrote: Perhaps stupid question but, is Asterisk a statefull or stateless proxy? Ignace _ Post Classifieds on MSN classifieds. http://go.msnserver.com/IN/44045.asp Buy and Sell on MSN Classifieds. ___ 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] Stripping numbers at the end of a dial pattern = extension
Hello,
From: Hermann Wecke [EMAIL PROTECTED]
Subject: [Asterisk-Users] Stripping numbers at the end of a dial pattern =
extensions.conf
Date: 8 May 2004 22:03:57 +
Is it possible to strip some numbers from the *end* of a number?
I know that ${EXTEN:1} will remove 1 position from the beggining... but
how to remove N numbers from the end?
Use ${variable:pos:n}. This will give you 'n' digits from the position
'pos'.
exten = 12345, 1, SetVar,MYDIGITS=${EXTEN:2,3} ; MYDIGITS = 2345.
Also, there is Substring application available with Asterisk, but it is
deprecated i think...
HTH, Girish
_
Post Classifieds on MSN classifieds. http://go.msnserver.com/IN/44045.asp
Buy and Sell on MSN Classifieds.
___
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] Stripping numbers at the end of a dial pattern = extension
Hi,
Replying to my own mail. There is a mistake, The syntax is incorrect:
From: Girish Gopinath [EMAIL PROTECTED]
exten = 12345, 1, SetVar,MYDIGITS=${EXTEN:2,3} ; MYDIGITS = 2345.
Correct: exten = 1234, 1, SetVar,MYDIGITS=${EXTEN:2:3} ; MYDIGITS = 234.
My apologies...
Girish
_
Sports, sports and more sports! Keep up with all thats happening!
http://www.msn.co.in/sports/ Stay connected with MSN Sports!
___
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] Ser and Asterisk together
Hello, From: Dawid Mielnik [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Ser and Asterisk together Date: Wed, 21 Apr 2004 21:15:21 +0200 We are using ser together with *. Ser is used as a SIP proxy/registrar, * is used as a sip - pstn gateway and voicemail/forward/conference server. Advanteges - scalable, very large number of sip clients with easier radius/database user management, advanced sip logic/routing options, better sip interoperability Our's is only a SIP based system and we use SER in front of Asterisk as SIP Proxy/Registrar. Asterisk is mainly used as a Media Server that plays IVR and for voice mails. Yes, it is highly scalable and database management is easy. We dont have SIP peers for Asterisk. It just plays the IVR and routes the call back to SER when it receives dtmf. As told by OEJ in a mail off the list, it is nothing but just a SIP call for Asterisk. Currently our system is under testing. disadvantages - you've got two boxes, no iax on ser so you still have to manage iax users on asterisk We are using only one box, SER is on 5060 and Asterisk on a different port. In my opinion, if you plan to deploy large number of sip clients - it's a good idea Very true. Dave Regards, Girish _ Marriage? http://www.bharatmatrimony.com/cgi-bin/bmclicks1.cgi?72 Join BharatMatrimony.com for free. ___ 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] reboots
Hello, That all being said, the machine has been running 174 days at this point. I recently crashed asterisk when trying to integrate a much newer version of asterisk into the IAX2 part of the network. Sorry, I did not understand. Trying to integrate into the IAX2 part of the network? Could you please elaborate that a bit? Regards, Girish _ Marriage? http://www.bharatmatrimony.com/cgi-bin/bmclicks1.cgi?72 Join BharatMatrimony.com for free. ___ 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] Help for Asterisk and kphone
Kiran, From: kiran p [EMAIL PROTECTED] Subject: [Asterisk-Users] Help for Asterisk and kphone hi Iam new to Voip and hence do not know much about Asterisk and Kphone,I need to install these for basic voip features between two computers, can anyone help me on where i can get started with this thanks kiran Find here: http://www.asteriskpbx.org/index.php?menu=support (See the User Contributed Links) http://www.voip-info.org/wiki-Asterisk Regards, Girish _ Marriage? http://www.bharatmatrimony.com/cgi-bin/bmclicks1.cgi?72 Join BharatMatrimony.com for free. ___ 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] Strange T1 Problem
Hello, From: Joe Dennick [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Strange T1 Problem Date: Fri, 16 Apr 2004 07:44:17 -0500 Can one use a pipe '|' for the Dial application the same way that one would use a comma ','? snipped I know this one works, but what I don't know is if it will also work using pipes in place of the commas. Joe Yes, You can use '|' for the dial application. In fact even if you use comma(,) in your extensions.conf, Asterisk replaces it with '|' when it builds the dial plan. See the following entry in extensions.conf: [test] exten = 1234,1,Dial(SIP/1234,20,r) exten = 1234,2,Voicemail(u1234) and the dialplan for this is: * CLI show dialplan test [ Context 'test' created by 'pbx_config' ] '1234' = 1. Dial(SIP/1234|20|r) [pbx_config] 2. Voicemail(u1234) [pbx_config] Regards, Girish _ Easiest Money Transfer to India. Send Money To 6000 Indian Towns. http://go.msnserver.com/IN/42198.asp Easiest Way To Send Money Home! ___ 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] Newbie....
Hello, From: Hall, Eric M. [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Newbie Date: Wed, 31 Mar 2004 22:18:56 -0500 Could I do things like call other ext on the system? Check Voice mail? I would like to test this before I put money in cards I may not need. What Software Phone app is people using? Yes, you can do such things. For configuring Asterisk and adding extensions, see the user contributed links here:http://www.asteriskpbx.org/index.php?menu=support You can use SJPhone, X-lite with windows. Linophone and LIPZ4 can be used with Linux. For using IAX protocol, use IAXPhone. Regards, Girish _ Get the best deals. On Electronics, Mobiles, Laptops. Log on to www.baazee.com http://go.msnserver.com/IN/45530.asp ___ 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 - SIP Interoperability
Hello,
From: pesb [EMAIL PROTECTED]
Subject: [Asterisk-Users] H323 - SIP Interoperability
Date: Thu, 1 Apr 2004 12:37:17 -0300
snip
So, I would like to call SIP/4 phone by dialing 014. Something like this:
exten = 01X,1,Dial(SIP/X) ; This is not working
How can I do that?
Try this:
exten = _01X,1,Dial(SIP/${EXTEN:2})
That should do it.
Another question: How can I make the RTP data flow go directly from one IP
phone to the other? Rigth now, all the RTP data flow goes through the SIP
proxy.
set canreinvite=yes for sip users in sip.conf
Regards, Girish
_
Easiest Money Transfer to India. Send Money To 6000 Indian Towns.
http://go.msnserver.com/IN/42198.asp Easiest Way To Send Money Home!
___
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 is the archive?
Hi, From: Matt Lawson [EMAIL PROTECTED] Subject: [Asterisk-Users] Where is the archive? Date: Thu, 01 Apr 2004 14:20:04 -0500 http://www.mail-archive.com/[EMAIL PROTECTED]/maillist.html only seems to go back a few days. Is there another archive somewhere that goes back farther? http://lists.digium.com/pipermail/asterisk-users/ Regards, Girish _ Contact brides grooms FREE! http://www.shaadi.com/ptnr.php?ptnr=hmltag Only on www.shaadi.com. Register now! ___ 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-msmessenger
Hello, From: Shawn [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Subject: [Asterisk-Users] sip-msmessenger Date: 31 Mar 2004 17:03:08 -0500 snip Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.100:9082 From: sip:[EMAIL PROTECTED];tag=97442d5b-75b7-4e23-9021-b8605797eb56 To: sip:[EMAIL PROTECTED];tag=as38587f8c Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=26e59066 Content-Length: 0 Just guessing here... Check your secret, host fields of the sip user entry in sip.conf. Or try removing the secret field and change host= field to host=dynamic. HTH... Regards, Girish _ Easiest Money Transfer to India. Send Money To 6000 Indian Towns. http://go.msnserver.com/IN/42198.asp Easiest Way To Send Money Home! ___ 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] LipZ4 Sip Soft Phone
Eliot, From: Eliot Robinson [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] LipZ4 Sip Soft Phone where can we find configuration for the LipZ4 Sip Soft Phone for * and how do you configure * for the LipZ4 Sip Soft Phone? Check out this url: http://www.zultys.com/products/lipz4/lipz4_quick_start.pdf This explains about configuring LipZ4 softphone. It is pretty simple. same questions for xten lite. Not sure, I haven't used xten lite. Regards, Girish _ Easiest Money Transfer to India. Send Money To 6000 Indian Towns. http://go.msnserver.com/IN/42198.asp Easiest Way To Send Money Home! ___ 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] LipZ4 Sip Soft Phone
Hi, From: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] LipZ4 Sip Soft Phone Date: Sat, 20 Mar 2004 09:33:42 -0500 (EST) Thanks a lot I might give it a try. Any specific instructions for running it with asterisk? AJ Checkout these urls, these might be of your interest: http://www.zultys.com/products/lipz4/softphone-1.3.11-0.i386.rpm http://www.zultys.com/products/lipz4/lipz4_quick_start.pdf http://www.zultys.com/download_manuals.htm Regards, Girish _ Protect your PC from viruses. Get in the experts. http://www.msn.co.in/pcsafety/ Click here now! ___ 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] Can i do voice chat without using the hardware
Suresh, From: suresh kumar [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Can i do voice chat without using the hardware Date: Fri, 19 Mar 2004 05:50:00 -0800 (PST) Thanks a lot for your valuable information. I will go through it once again. Still i don't have any idea to connect two PC's. Hope i may get help from you. For configuring 2 softphones with Asterisk see this link: http://www.automated.it/guidetoasterisk.htm That helped me a lot in learning Asterisk. It explains configuring your sip phones with Asterisk. Is there any softwares like X-lite for Linux? Yes, I think you can use linophone. But i was not able to install linophone because of some make issues. Also i have tested the softphone from zultys. It works well with Asterisk. You can get it from their web site:http://www.zultys.com Regards, Girish _ Catch the formula fever! Get all the latest news. http://www.msn.co.in/formula2004/ Right here on MSN. ___ 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] LipZ4 Sip Soft Phone
Hi, From: [EMAIL PROTECTED] Subject: [Asterisk-Users] LipZ4 Sip Soft Phone Date: Fri, 19 Mar 2004 19:54:58 -0500 (EST) Hello all, In browsing through my latest version of Linux Journal I noticed the LipZ4 Sip Soft Phone advertised. I was wondering if anyone had yet used it with asterisk? And if so, what were the results? Thanks. AJ Yes, i have tested that with Asterisk, and it works pretty good. User interface is also good. But the only problem i face is, once i close the application, it doesn't get loaded next time. I have to restart my linux box to get it working again. Regards, Girish _ Easiest Money Transfer to India. Send Money To 6000 Indian Towns. http://go.msnserver.com/IN/42198.asp Easiest Way To Send Money Home! ___ 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] Can i do voice chat without using the hardware
Suresh, From: suresh kumar [EMAIL PROTECTED] Subject: [Asterisk-Users] Can i do voice chat without using the hardware snip Without using any of Digium's hardware or T1 or E1 interfaces , can i do voice chat between two computers (intranet/internet)? If possible, How can i do that? (Any configuaration setting is required?) Sur You can use Asterisk's SIP channels for this. You need to have IP phones/Softphones that register with Asterisk. If you are using windows you can freely download and install evaluation versions of SJphone or Xlite. For configurations, see the 'User Contributed Links' section of http://www.asteriskpbx.org/index.php?menu=support There are plenty of documents available to help you... Good luck, Girish _ Take a loan. Win great prizes! Handsome prizes to be won! Take a loan win TV, Fridge many more prizes ! http://go.msnserver.com/IN/44044.asp ___ 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] Calling one local SIP user from another (using X-Lite)
Ross, From: Ross Finlayson [EMAIL PROTECTED] Subject: [Asterisk-Users] Calling one local SIP user from another (using X-Lite) Date: Sun, 14 Mar 2004 18:21:20 -0800 snip voice mail. However, if I try to call user2 from user1's X-Lite - or vice-versa - I get a 404 Not Found error. Is there anything obvious that I'm doing wrong? (In particular, do I also need to add entries to extensions.conf for user1 and user2??) Ross. Try adding something similer to this in the default context of your extensions.conf: exten = your exten,1,Dial(SIP/user1,20, tr) exten = your exten,1,Dial(SIP/user2,20, tr) Regards, Girish _ Take a loan. Win great prizes! Handsome prizes to be won! Take a loan win TV, Fridge many more prizes ! http://go.msnserver.com/IN/44044.asp ___ 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] need advice (newbies)
Hello, From: Nil MEKKI [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] need advice (newbies) Can anyone answer please ? See the information here: http://www.voip-info.org/tiki-index.php?page=Asterisk%20hardware%20recommendations Also, please do some googling and research before asking your questions. There are many resources available to help you configuring and using asterisk. I would like to know what hardware I need to set up my VOIP/PBX system ! I have 4 T1/E1 phone lines. If I choose Digium for instance ! Which cards do I need to buy to start and how many of them ? Regards, Girish _ INDIA TODAY @ Rs. 5 + a free gift ! http://www.indiatoday.com/itoday/intlsubscription/itsubs/it_offer.html Subcribe Now ... ___ 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: RE[2]: [Asterisk-Users] Crossconnect VoIP and PSTN in India. Is it allowed?
Hello, From: Vasyl Rublyov [EMAIL PROTECTED] But what about if we have Lucent Definity already in place and I would like to connect to Definity using PRI Asterisk box and route all USA and Ukrainian extensions thru VoIP? The same for people in US/UA - they will dial Indian extensions and go thru, does it possible/permitted? The PBX (Definity) already connected to PSTN in India. Not much idea about that, but i heard that the rates are fixed by the MTNL. What about those providers? http://www.voipproviderslist.com/details/countries/132 Again Sorry, :( Also I see those topics... did I miss something? http://www.zdnetindia.com/news/specials/voip/stories/53302.html http://www.apnic.net/mailing-lists/s-asia-it/archive/2002/03/msg00032.html This may be of interest :http://www.trai.gov.in/IP_Recommendations.htm But it takes ages to load... Regards, Vasyl Sorry for not having precise information about your queries, but i too was not able to find them. Good Luck, Girish _ Frustrated with dial-up? Lightning-fast Internet access for as low as $29.95/month. http://click.atdmt.com/AVE/go/onm00200360ave/direct/01/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Crossconnect VoIP and PSTN in India. Is it allowed?
Hello, From: Vasyl Rublyov [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Crossconnect VoIP and PSTN in India. Is it allowed? Date: Sun, 07 Mar 2004 11:22:53 -0500 Hello All, Does anyone know if it possible to crossconnect PSTN and VoIP system in India? AFAIK, It is not permitted here in INDIA. The Telecom Department (BSNL) does not allow to use their infrastructure to connect with VoIP systems. But connecting a PC to PC, and PC to PSTN Phone, where PC in India and Phone abroad is allowed. The recommendations submitted by the Telecom Regulatory Authority to the Govt, defines internet telephony as an application service. See here:http://www.zdnetindia.com/news/specials/voip/stories/50937.html Regards, Girish _ Contact brides grooms FREE! http://www.shaadi.com/ptnr.php?ptnr=hmltag Only on www.shaadi.com. Register now! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] re: cdr and macros
Yair,
As stated by OEJ in one of his old posts, using 's' is not a good thing for
CDRs. Instead, you can try something like this to get the correct CDR.
exten = _XXX,1,Macro(extensip,${EXTEN})
[macro-extensip]
exten = s,1,goto(dialout,${ARGV1})
[dialout]
exten = _X.,1,Dial(SIP/extension,10)
Regards, Girish
From: yair hakak [EMAIL PROTECTED]
i've been playing with cdr_mysql and the Master.csv file, and since i use a
macro to define extensions the csv and the db both save the destination of
the call as s, instead of the destination.
macro is as follows:
[macro-extensip]
exten =s,1,Dial(SIP/${ARG1},10)
exten =s,2,Voicemail(u${ARG1})
exten =s,3,Hangup()
exten =s,102,Voicemail(b${ARG1})
exten =s,103,Hangup()
and extension matching:
exten = _XXX,1,Macro(extensip,${EXTEN})
pretty standard stuff. how can i get the cdr to show the actual
destination? I guess i could parse the dstchannel field but i'd rather see
what the user actually dialed as well.
sorry if i'm missing something basic,
yair
_
Contact brides grooms FREE! http://www.shaadi.com/ptnr.php?ptnr=hmltag
Only on www.shaadi.com. Register now!
___
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] Caller ID Name Display
Hi,
But what I can't figure out is how to pass the CallerID Name instead of the
CallerID number?
use this: ${CALLERIDNAME}
Find more info at: /usr/src/asterisk/doc/README.variables
Regards, Girish
_
Need a job? Get head-hunted by the best.
http://www.naukri.com/msn/index.php?source=hotmail Post your CV free!
___
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] canreinvite=yes in sip.conf still go through asterisk
Zen, I am trying to confirm the command 'canreinvite=yes' in sip.conf using grandstream BT101/2s and snom100s. In either case, no description nor 'canreinvite=yes', media stream always go through *. Do I need another settings for confirming sip clients directly communicate each other? Do you have a Dial statement that has t or T in it? This will force the media stream to pass through Asterisk. Regards, Girish _ Contact brides grooms FREE! http://www.shaadi.com/ptnr.php?ptnr=hmltag Only on www.shaadi.com. Register now! ___ 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] DTMF Issues with SJPHONE
What ver of SJPHONE? SJPhone Evaluation Version, release Jul 31, 2003, Build: 1.10.187c Girish _ All the news that matters. All the gossip from home. http://www.msn.co.in/NRI/ Specially for NRIs! ___ 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] DTMF Issues with SJPHONE
Has anyone had a similar issue with Asterisk Voicemail being unable to detect the digits sent from an SJ Phone connection. I have included dtmfmode=inband and it works fine when calling other phones though not with Voicemail. Voicemail doesn't regonise the password. I am using SJPhone, and works fine for me. Is there a way to not send a password when logging into Voicemail as a temp measure. Try something like like this, it will not ask for your password: exten = your extension,1,Ringing exten = your extension,2,Wait(2) exten = your extension,3,VoicemailMain,s ; is the mail box number Also, check out this url: http://www.automated.it/guidetoasterisk.htm Regards, Girish _ Post Classifieds on MSN classifieds. http://www.sulekha.com/msnclassifieds Buy and Sell on MSN Classifieds. ___ 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] RE: sip router gateway
Anand, My question: CAN I use asterisk software as a replacement for siprg ? CAN I use DIGIUMs x100p PCI card as a replacement for QUICKNETS CTI LINE JACK CARD ? Asterisk is a soft PBX that has a SIP channel. It doesnt perform all the functionalities of a sip router. You can use digium's cards with asterisk, but is it legal to use such cards here in INDIA? As far as i know BSNL relegations dont permit to use their infracture to use with VoIP. They dont allow to land VoIP calls to a PSTN phone/mobile. (But it is possible, illegal though) Yes, since we are doing this as part of the educational programme, and more over we plan to take special permission from BSNL towards this. Could you please explain me how does this soft PBX works ? and I want all the functionalities of the sip. Please see here: http://www.asteriskpbx.org/index.php?menu=features. Friends, correct me if i am wrong: Unlike a sip proxy that routes your sip requests/ responses, asterisk forces the rtp media stream to pass through it. This is required for the features like call transfer etc. But there are provisions to disable this feature. Asterisk has a PBX core whereas SIP proxies have their proxy core and they normally dont generate '100 Trying' responses for non INVITE requests. Asterisk can be used with PSTN as well as VoIP. It provides all telephony services. All the functionalities of SIP, not sure, At least my installation does not support SUBSCRIBE, NOTIFY methods. CAN ASTERISK work as SIP ROUTER GATEWAY ? Asterisk is a PBX, If you just want a SIP router, i think it is better to use SIP Express Router or anything like that. but sip express router, does it supoort linphone,gnuosip,partysip? and what linejack card should i be using for this ? Not sure. Others here must be knowing about these issues. SER, partysip etc work with asterisk Also i dont have knowledge about linejack cards etc. CAN I USE ASTERISK, DIGIUMs card along with LINPHONE, PARTYSIP and GNUoSIP. ? Partysip and Linphone uses oSIP. right? Asterisk does not use any external Yes protocol stacks. It has its own implementation. I am not sure, but are you planning to use * and Partysip in a same machine? What about SIP ports? I could be wrong. Yes, want to use it on the same machine,as you said, i really dont know whether the asterisk sip stack and gnuosip will collide..definetly. I have a RH9 box installed with * and Partysip. But i have never tried to run them together. Regards, Girish _ Post Classifieds on MSN classifieds. http://www.sulekha.com/msnclassifieds Buy and Sell on MSN Classifieds. ___ 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] cdr-dst incorrect?
Sam,
From an OEJ Post:
Try to match every extension before dialing out instead, using s is a bad
thing for CDRs.
[default]
exten = 1234,1,macro(dial-out)
[macro-dial-out]
exten = s,1,Dial(SIP/test,30,r)
[default]
exten = 1234,1,macro(dial-out,${EXTEN})
[macro-dial-out]
exten = s,1,goto(dial-out2,${ARGV1})
[dial-out2]
exten = _X.,1,Dial(SIP/test,30,r)
Of course, you could to a goto instead of macro in the first place, but
there might be
another reason that you want to use a macro...
**
http://lists.digium.com/pipermail/asterisk-users/2004-January/033970.html
Hope it solves your problem...
Regards, Girish
From: SamW [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] cdr-dst incorrect?
Date: Tue, 24 Feb 2004 23:58:40 -0500
I am using following setup to dialout, I can take calls through sip-out
which is defined in sip.conf. My issue is cdr records will have a s for
destination. What Can be wrong and any suggestions to fix? Can this be a
bug I am using Version 0.7.2.
instead of macro-dialout if I directly dialed through the [intern] I get
the correct results. Some how asterisk think I dialed extension s instead
of the number I dialed.
Any help appreciated.
[macro-dialout]
exten = s,1,Dial(SIP/[EMAIL PROTECTED],50)
exten = s,2,Busy
exten = s,102,Busy
[intern]
exten = _1NXXNXX,1,Macro(dialout,${EXTEN},60)
- SamW
_
All the news that matters. All the gossip from home.
http://www.msn.co.in/NRI/ Specially for NRIs!
___
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] softphone configs?
Feb 18 11:30:16 NOTICE[1125329600]: chan_sip.c:5577 handle_request: Registration from 'Mark sip:[EMAIL PROTECTED]' failed for '192.168.5.64' Pls check your 'username' and 'secret ' entries in your sip.conf (or remove those entries). I'm using SJPhone here. Girish _ Contact brides grooms FREE! http://www.shaadi.com/ptnr.php?ptnr=hmltag Only on www.shaadi.com. Register now! ___ 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] softphone configs?
Mark, Try with this config: [phone1] type=friend username=mark secret=elihu context=testsip callerid=Mark Messmore Some comments about ur sip.conf reinvite=no. Do not use this, instead use canreinvite check the spelling of callerid sjphone supports only inband dtmf Girish From: Mark Messmore, Technical Support, University Telcom Inc. [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] softphone configs? Date: Wed, 18 Feb 2004 14:56:41 -0500 k...Here you go. (that is if attachments are allowed). If not I'll find out in a minute and just send the text of the config. Thanks Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rana Dutt Sent: Wednesday, February 18, 2004 1:55 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] softphone configs? What does your sip.conf look like? Please include it in your next message in its entirety. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Messmore, Technical Support, University Telcom Inc. Sent: Wednesday, February 18, 2004 1:08 PM To: [EMAIL PROTECTED] Subject:[Asterisk-Users] softphone configs? I've tried using the x-lite softphone as well as sjphone. I've gone over my configurations a dozen times...and I always seem to get the following error: Feb 18 11:30:16 NOTICE[1125329600]: chan_sip.c:5577 handle_request: Registration from 'Mark sip:[EMAIL PROTECTED]' failed for '192.168.5.64' FYI...I'm trying to do all my voip internally, nothing to the outside world yet. If anyone could give me an idea I'd appreciate it. Thanks. Mark ___ 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 sip.conf _ Post Classifieds on MSN classifieds. http://www.sulekha.com/msnclassifieds Buy and Sell on MSN Classifieds. ___ 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] Is there a way to transfer a call from CLI
I think there is an option. But i dont know how to use it. When i enter the command 'show application transfer' on the CLI , it gives this: [Synopsis]: Transfer caller to remote extension [Description]: Transfer(exten): Requests the remote caller be transferred to. But when i enter 'help' on the CLI , Asterisk says: [snip] transfer Transfer a call to a different extension [snip] What does Asterisk transfer? The caller or the call? Or am i wrong here? I tried with 2 SIP soft phones. But it did not work. Asterisk said: 'There is no call to transfer' How can i acheive this feature? Regards... Girish From: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Is there a way to transfer a call from CLI Date: Mon, 26 Jan 2004 19:36:44 -0500 (EST) Does anyone know of a way to transfer a call from the CLI? AJ ___ 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 _ Games, MMS cards, ringtones. Operator logos, picture messages more. http://server1.msn.co.in/sp03/mobilesms/ Jazz up your mobile! ___ 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] Has Nufone gone belly-up
And I don't care about your network, your services, or your contributions to Asterisk. Your behaviour in this matter is like that of a toddler in a sandpit, throwing sand back at the other kids then screaming they started it. Grow up. Your prospective customers have. echo nufone.net killfile But I respect him. I have never seen him or he has never answered my doubts about Asterisk. Still i do respect him because of his contributions to Asterisk. And I respect all who contribute to Opensource. Even Mahatma Gandhi and Albert Einstein had problems in their character. Don't we respect them? Please stop this thread. It is not talking about Asterisk or Nufone. It just says about people's attitudes. Girish _ Add glamour to your desktop. Let your screen sizzle. http://server1.msn.co.in/msnchannels/Entertainment/wallpaperhome.asp Download the hottest wallpapers. ___ 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] Re: SayDigits
Chris,
May be your callerID contains characters that cannot be played by Asterisk.
For example:
if the callerid is 1010, Asterisk will not be able to find and play the
file '' in digits directory.
In your case i guess the caller id starts with
Regards...
Girish
From: Doug Meredith [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: SayDigits
Date: Sat, 24 Jan 2004 20:32:42 -0400
Chris Wilson [EMAIL PROTECTED] wrote:
Has anyone had this problem:
(When calling to ext. 1010)
Jan 24 10:50:27 WARNING[-1252262992]: file.c:446 ast_openstream: File
digits/ does not exist in any format
Jan 24 10:50:27 WARNING[-1252262992]: file.c:734 ast_streamfile: Unable
to open digits/ (format ULAW): No such file or directory
in Extensions.conf
exten = 1010,1,SayDigits(${CALLERID})
/var/lib/asterisk/sounds/digits exists, and there are many files in
there. Any idea's?
File permissions?
Doug
--
Doug Meredith ([EMAIL PROTECTED])
SystemGuard - Oracle remote support
877-974-8273 (87-SYSGUARD)
506-854-7997
www.systemguard.com
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_
Contact brides grooms FREE! http://www.shaadi.com/ptnr.php?ptnr=hmltag
Only on www.shaadi.com. Register now!
___
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] Searching the archives - new engine demo
Hi, Very good! I tried some of the features and they are really good, especially the search within selected months. I have been reading this list for the last 4 months. Answers for some(all) of my doubts are there in the mails posted within these months. Normally when googling, I try different combinations to find out these mails. It is ok, but time consuming. Your search engine is just perfect for people like me and i hope someone will host it. Regards... Girish From: Kim Hendrikse [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Searching the archives - new engine demo Date: Sat, 24 Jan 2004 13:21:13 +0100 Hi, I've placed a demo search engine of the asterisk users archive here: http://asterisk.nextrieve.com/cgi-bin/asterisk I know there are a number of other ways to search this list that have been suggested and one person suggested that another wasn't necessary but this engine will do some things that the others can't. Specifically you can search within selected months and from specific users. Or require a certain word in a subject. You also do a fuzzy search if you are not sure of the spelling, which approximates a phrase search. I'll leave this demo up for a couple of days, if this is interesting to people and someone wishes to host it I can provide the code. I've noticed myself that it can be difficult to search the list within certain time periods. Google simply won't do that. Not base upon the time the mail was sent. Make sure you check out the advance features which is where you can restrict to sender, exclude words etc. - Kim Hendrikse ___ 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 _ Rethink your business approach for the new year with the helpful tips here. http://special.msn.com/bcentral/prep04.armx ___ 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] Warning /:Asterisk.c:255 Listener : Select Returned Error
I am also getting this warning. But i noticed that this happens only when i issue a shell command from the Asterisk CLI. But it has never affected any functionality. My linux box is a very slow one. Regards... Girish From: Frankie Gravato [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: Asterisk [EMAIL PROTECTED] Subject: [Asterisk-Users] Warning /:Asterisk.c:255 Listener : Select Returned Error Date: Sat, 24 Jan 2004 14:49:05 -0500 Hello Asterisk-List Folks Jan 24 14:39:27 WARNING[1082805040]: asterisk.c:255 listener: Select retured error: Interrupted system call Jan 24 14:39:27 WARNING[1082805040]: asterisk.c:255 listener: Select retured error: Interrupted system call == Detected 4 licensed G.729 transcoders Jan 24 14:39:27 WARNING[1074411264]: translate.c:219 calc_cost: Tra Has any one ever seen this before i started getting this recently after i install my g729 codecs is there something wrong with my asterisk that its telling me cause i dont understand whats its asking me , -- Best regards, Frankie ([EMAIL PROTECTED]) mailto:[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 _ Marriage? Join BharatMatrimony.com for free. http://www.bharatmatrimony.com/cgi-bin/bmclicks1.cgi?74 ___ 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] exten=h and ResetCDR
Hi friends, I have the entry exten = h,Hangup in my extensions.conf, and I am trying to record the call details for billing. From the wiki i found out that the use of exten=h,... is not suggested for the CDRs. What impact will the use of 'h' make on CDRs? Also, what is the advantage of using ResetCDR with exten=h? Regards... Girish _ Easiest Money Transfer to India. http://go.msnserver.com/IN/41490.asp Send Money To 6000 Indian Towns. ___ 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] Starting with MGCP and Asterisk
Ricardo, I think that maybe the asterisk.conf file is missing?.. where i canf find a sample for this file? Run: make samples Did you read the message displayed by Makefile after installing Asterisk? Girish _ Add glamour to your desktop. Let your screen sizzle. http://server1.msn.co.in/msnchannels/Entertainment/wallpaperhome.asp Download the hottest wallpapers. ___ 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] Parking extension:700
Hi all, From Andy Powells Getting Started With Asterisk (V 0.1a) http://www.automated.it/guidetoasterisk.htm parking.conf file has this number set at 700. I've changed mine to 701 because I was having an issue with Asterisk - although it would 'see' (looking at the console) I had tried to transfer to 700 it appeared not to believe that I had dialed it. This was essentially due to the 00 in the 700, changing it to 701 eliminates the problem completely. The extension 700 is working for me. I am able to park the call. I am using SJPhone here. As stated in that guide Is there any problem with using 00? If yes, can anyone explain that? Regards... Girish _ Marriage? http://www.bharatmatrimony.com/cgi-bin/bmclicks1.cgi?74 Join BharatMatrimony.com and get married. ___ 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: Re: [Asterisk-Users] How to park and pickup a call
By the way, I have an other question, are there any way to implement MeetMe conference If I haven't zaptel device? Zhang Peihao 2004-01-15 Yes, there is. Modify the Makefile in the /usr/src/zaptel directory, ie, change #ztdummy to ztdummy and run make clean; make install Find more information at: http://www.automated.it/guidetoasterisk.htm Regards... Girish _ Get head-hunted by 10,000 recruiters. http://go.msnserver.com/IN/35984.asp Post your CV on naukri.com today. ___ 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] call parking
Sean, Check out this url: http://www.automated.it/guidetoasterisk.htm Girish From: Sean Garland [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] call parking Date: Tue, 13 Jan 2004 19:41:05 -0800 I am having trouble with call parking... I am basically using the stock sample files, but extension 700 doesn't show up in my dialplan. When I transfer a call to 700, I get the fast busy like there is extension 700... HELP! Sean Garland _ Free transactions in any ATM across India. http://server1.msn.co.in/msnleads/suvidha/dec03.asp?type=hottag Click here. ___ 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] Problems compiling cdr_pgsql
Hi, Having installed postgresql-devel-7.4-0.3 and postgresql-libs-7.4-0.3 I'm having probs. compiling cdr_pgsql, can anyone offer any pointers as to what I might be missing? I'm hoping I've just missed out something like postgresql-wibblewobble-7.4-0.3 or something ... Below is the result of a make in the cdr source dir which may help those of you in the know thanks... Andy [EMAIL PROTECTED] cdr]# make Run make install from /usr/src/asterisk directory Girish _ Send DD, pay no commission. http://server1.msn.co.in/msnleads/suvidha/dec03.asp?type=hottag Click here. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * Stresstool Help required
Alastair, You were correct. The program was generating the same call IDs for all the INVITEs. In fact it is a small routine of just 2 lines. I checked all my routines, except this one. I never expected a bug in such a small routine. It taught me a good lesson. Sorry, i was not able to post this mail yesterday due to some network problems here. Thanks for helping me out... Warm Regards, Girish From: Alastair Maw [EMAIL PROTECTED] On 02/01/04 14:24, Girish Gopinath wrote: I gave the sip debug command, and one of the lines showed:Ignoring this request Can you log the SIP debug messages to a file and put it up on the web somewhere? Or do an ethereal capture or similar. It's very hard to say what the problem might be without a full SIP trace. It's likely that you're generating the same transaction ID for each SIP INVITE or something silly. Regards, Alastair _ Free transactions in any ATM across India. http://server1.msn.co.in/msnleads/suvidha/dec03.asp?type=hottag Click here. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * Stresstool Help required
Alastair, You were correct. The program was generating the same call IDs for all the INVITEs. In fact it is a small routine of just 2 lines. I checked all my routines, except this one. I never expected a bug in such a small routine. It taught me a good lesson. Sorry, i was not able to post this mail yesterday due to some network problems here. Thanks for helping me out... Warm Regards, Girish From: Alastair Maw [EMAIL PROTECTED] On 02/01/04 14:24, Girish Gopinath wrote: I gave the sip debug command, and one of the lines showed:Ignoring this request Can you log the SIP debug messages to a file and put it up on the web somewhere? Or do an ethereal capture or similar. It's very hard to say what the problem might be without a full SIP trace. It's likely that you're generating the same transaction ID for each SIP INVITE or something silly. Regards, Alastair _ Start getting interview calls immediately. http://go.msnserver.com/IN/40245.asp Post your CV on naukri.com today. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * Stresstool Help required
Hi all, I am trying to write a program that sends SIP requests to asterisk. My aim is to make asterisk record as many voicemails it can at a time. The design of the program is like this: There are two processes: One main process and a child process (No flames pls. I have very little idea about pthreads and dl modules) The main program asks the user to input the number of test instances. When the user inputs that (valid instances are: 1 - 50), it will spawn that many number of child processes that communicate with *. All of them get their own sip ports, rtp ports, and user names (for REGISTERing with *). There is a delay of 4 seconds before spawning each process. When i input 1, everything works fine (i guess). * records the voicemail (i am sending the contents of a .wav file to asterisk) . Here is the screen capture: *CLI -- Registered SIP 'gopi' at 192.168.68.15 port 5061 expires 120 == Setting SIPDOMAIN to : 192.168.68.6 -- Executing Dial(SIP/gopi-bddf, SIP/stest|10|tr) in new stack == Everyone is busy at this time -- Executing Ringing(SIP/gopi-bddf, ) in new stack -- Executing Answer(SIP/gopi-bddf, ) in new stack -- Executing VoiceMail2(SIP/gopi-bddf, u) in new stack -- Playing 'voicemail/default//unavail' (language 'en') -- Playing 'vm-intro' (language 'en') -- Playing 'beep' (language 'en') WARNING[15376]: File app_voicemail.c, Line 1236 (leave_voicemail): No more messages possible -- Executing Hangup(SIP/gopi-bddf, ) in new stack == Spawn extension (stresstest, , 5) exited non-zero on 'SIP/gopi-bddf' The problem starts when i try to spawn more than one instance of the process. I tried with 2, both the instances got registered. The initial part of dialing is also ok. After that one of the child processes gets BYE request from *. The other child continues and * records voicemail for it. Here is the screen capture of that: -- Registered SIP 'gopi' at 192.168.68.15 port 5061 expires 120 == Setting SIPDOMAIN to : 192.168.68.6 -- Executing Dial(SIP/gopi-7263, SIP/stest|10|tr) in new stack == Everyone is busy at this time -- Executing Ringing(SIP/gopi-7263, ) in new stack -- Executing Answer(SIP/gopi-7263, ) in new stack -- Executing VoiceMail2(SIP/gopi-7263, u) in new stack -- Playing 'voicemail/default//unavail' (language 'en') -- Registered SIP 'nath' at 192.168.68.15 port 5062 expires 120 == Setting SIPDOMAIN to : 192.168.68.6 -- Executing Dial(SIP/nath-bedf, SIP/stest|10|tr) in new stack == Everyone is busy at this time -- Executing Ringing(SIP/nath-bedf, ) in new stack -- Executing Answer(SIP/nath-bedf, ) in new stack -- Executing VoiceMail2(SIP/nath-bedf, u) in new stack -- Playing 'voicemail/default//unavail' (language 'en') -- Playing 'vm-intro' (language 'en') -- Playing 'vm-intro' (language 'en') WARNING[5126]: File chan_sip.c, Line 469 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 2 (Response) WARNING[16400]: File file.c, Line 512 (ast_readaudio_callback): Failed to write frame == Spawn extension (stresstest, , 4) exited non-zero on 'SIP/gopi-7263' -- Playing 'beep' (language 'en') WARNING[17425]: File app_voicemail.c, Line 1236 (leave_voicemail): No more messages possible -- Executing Hangup(SIP/nath-bedf, ) in new stack == Spawn extension (stresstest, , 5) exited non-zero on 'SIP/nath-bedf' I gave the sip debug command, and one of the lines showed:Ignoring this request Does that mean asterisk doesn't process 2 requests simultaneously, when it is sent from one machine? I know * is not a SIP proxy, it is a PBX. Is this problem related to that? If so, how * registered the two instances of the process? I tried with one instance of the test program from one machine, and SJPhone from another machine. Both worked fine. Can anybody help me in figuring out the problem? I admit that there are many bugs in my program and i beleive that the problems are because of these bugs only. Still wanted to hear from you... Warm Regards... Girish _ Gujarat Kite Fest at http://go.msnserver.com/IN/40247.asp www.gujaratkitefest.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New to asterisk? RUN... don't walk.
Excellent!!! Well Said, JR... From: JR Richardson [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] New to asterisk? RUN... don't walk. Date: Thu, 1 Jan 2004 10:11:36 -0600 Piping in 2 cents, This is a great example of the Internet, Fast Food generation, showing their appreciation for all the magic that happens in the labs, hearts and minds of the courageous, hard working, dedicated and motivated group of people truly interested and guided to accomplish greatness. This platform for learning is one of the best tools in existence to come to a finite understanding of VoIP and legacy telephony with the versatility to expand beyond and develop originality in the field of telecommunications excellence, product development. Learn it, understand it, appreciate it, then take it past where you found it and if you're capable contribute, if not, enjoy it. But always, always maintain respect for those who created it and continue to refine it. Learning is intrinsically human, and in this world of Industry (There is no substitution for knowledge. [Edward Deming]). Find your inner child, re-capture and embrace what God has given you, the ability to learn. It will require you to put down the remote control, get off the couch and decrease your apparently frequent visits to McDonalds. Search and find the knowledge which you seek to ultimately fulfill your destiny; build an Asterisk Server that works. Hell, we all did. JR 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 _ Send DD, pay no commission. http://server1.msn.co.in/msnleads/suvidha/dec03.asp?type=hottag Click here. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SJphone, Asterisk and DTMF tones ...
Hi, I am using SJPhone here for testing ivr with Asterisk. I haven't seen any problem here yet. I have tried different things for that and finally got it working. I am not an expert to explain more about that, but here is the general section form my sip.conf. dont know whether it will help... [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls allow=ilbc|gsm|ulaw|g723.1|g711 ;allow=all dtmfmode=inband ;dtmfmode=inband|rfc2833 good luck... Girish From: Darren Nickerson [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SJphone, Asterisk and DTMF tones ... Date: Sun, 21 Dec 2003 01:29:16 -0500 Folks, I can't seem to get DTMF signaling working properly using SJphone connecting to Asterisk via a SIP connection. Here's an example of a voicemail session where I entered 1234 for both the username and the password: -- Incorrect password '11223344' for user '11223f344' (context = any) This is with dtmfmode=inband in sip.conf. With either rfc2833 or info, DTMF tones don't seem to get 'seen' by Asterisk at all. I'm running CVS-12/17/03-02:39:14, in case it's relevant. Help? -Darren -- Darren Nickerson Senior Sales Support Engineer iFax Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 office +1.215.243.8335 fax ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users _ Add glamour to your desktop. Let your screen sizzle. http://server1.msn.co.in/msnchannels/Entertainment/wallpaperhome.asp Download the hottest wallpapers. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IVR sample config?
exten = _X., Goto(ivrmenu,s,1) [ivrmenu] exten = s,1,Ringing exten = s,2,DigitTimeout,30 exten = s,3,Background(something) ; press 1 for sales exten = t,3,Goto(business,0,1) include = business ;map yourl extens here... [business] exten = 0,1,... exten = 1,1, .. From: Rich Adamson [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: Asterisk-a-users-list [EMAIL PROTECTED] Subject: [Asterisk-Users] IVR sample config? Date: Sat, 20 Dec 2003 08:37:26 -0600 Can someone point me to some reasonable example / starting point to implement a basic IVR menu? Looking for something rather simple like the press 1 for sales, 2 for tech support, and probably an option to list the voicemail directory kind of thing. Nothing elaborate needed, just basic menu. (Yes, I did look at the wiki and google searched for ivr menu.) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users _ Dont miss out on jobs that are not advertised. http://go.msnserver.com/IN/38902.asp Post your CV on naukri.com today. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Meetme Recording
Hi, Can anybody explain me in configuring Asterisk to record a conference? Regards... Girish _ Add zing to Hotmail. Get FREE newsletters. http://server1.msn.co.in/features/general/Newsletters/index.asp Subscribe now! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Meetme Recording
Hi Gus, Thanks. It worked regards... Girish From: CW_ASN - Gus [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Meetme Recording Date: Tue, 2 Dec 2003 10:56:18 -0300 Try something like this: exten = 2060,1,Answer exten = 2060,2,Wait,1 exten = 2060,3,Monitor,wav|algo exten = 2060,4,Meetme,1|ps Regards, Gus - Original Message - From: Girish Gopinath [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, December 02, 2003 8:58 AM Subject: [Asterisk-Users] Meetme Recording Hi, Can anybody explain me in configuring Asterisk to record a conference? Regards... Girish _ Add zing to Hotmail. Get FREE newsletters. http://server1.msn.co.in/features/general/Newsletters/index.asp Subscribe now! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users _ Download cool KHNH ringtones. Add style to your mobile. http://server1.msn.co.in/sp03/gprs/howcani_ring.asp Simply click here. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Meetme Recording
Dave, 'algo' is the file in which u store the recorded conversation. u can find this file in /var/spool/asterisk/monitor Girish From: Dave Packham [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED], [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Meetme Recording Date: Tue, 02 Dec 2003 09:59:14 -0700 what do the options algo do in the monitor app? I dont see that in the show application monitor? is this a patch? Dave [EMAIL PROTECTED] 12/2/2003 6:56:18 AM Try something like this: exten = 2060,1,Answer exten = 2060,2,Wait,1 exten = 2060,3,Monitor,wav|algo exten = 2060,4,Meetme,1|ps Regards, Gus - Original Message - From: Girish Gopinath [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, December 02, 2003 8:58 AM Subject: [Asterisk-Users] Meetme Recording Hi, Can anybody explain me in configuring Asterisk to record a conference? Regards... Girish _ Add zing to Hotmail. Get FREE newsletters. http://server1.msn.co.in/features/general/Newsletters/index.asp Subscribe now! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users _ Add zing to Hotmail. Get FREE newsletters. http://server1.msn.co.in/features/general/Newsletters/index.asp Subscribe now! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Virtual PBX (*)
Hi, I have received some replies for my previous mail (* configuration), asking for my goals in configuring Asterisk. So here they are: We are planning to host an Inter continental virtual PBX service that will enable our users to register for an account which give them a toll-free # or a DID. Once registered using a web based interface, that user can add as manay extensions he/she wants and people can be conencted from remote locations in Japan, India using Asterisk and have a US based phone #. We already have tie-ups with some of the carrier providers in the US. The initial tests were successful, we were able to divert calls using IAX2, and we are expecting around 100 virtual PBX owners and around 2000 users to make use of this facility, that includes features like Call forwarding, VM, IVR, NAT traversal etc. I'm trying to put together a document for my company which would help users in setting a service of this magnitude using Asterisk and help here would be greatly accepted, and I promise once finished I will publish all my test results here in the group. Now, lets assume I have a linux box on a dual processor 3.2 GHz Intel box with 4GB RAM, RAID system and at the data center we would have bandwidth of around 1Mbit 1) How many Voicemails can be recorded at a time? 2)How many IVR's it can handle simultaneously? 3)Whether Postgres or Mysql is best suited? 4)How many NAT traversal relay sessions can be there at a time? 5)How to cluster multiple * boxes? (for failover dialing) I would also appreciate any user experience in setting up similer systems using *. and the problems they had to face etc... Regards... Girish _ BharatMatrimony.com. http://www.bharatmatrimony.com/cgi-bin/bmclicks1.cgi?74 India's premium matrimonial website. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] Virtual PBX (*)
Hi WipeOut, Thanks for the information provided. I would be greatful if u can clarify the following doubts: Girish Gopinath wrote: .. .. Now, lets assume I have a linux box on a dual processor 3.2 GHz Intel box with 4GB RAM, RAID system and at the data center we would have bandwidth of around 1Mbit 2000 users on 1Mbps?? I think you will probably need more bandwidth.. Using iLBC, 1Mbps will only give you about 40 concurrent VoIP channels.. *** Sorry! That's a mistake. In fact bandwidth is not a problem *** 1) How many Voicemails can be recorded at a time? Depends on where the call is coming from (PSTN via Tx00P or VoIP) and what format you are storing voicemails in (wav or gsm).. *** What difference does it make for PSTN and VoIP? *** Any OpenSource Tools available for testing these? Regards... Girish _ 10,000 recruiters head-hunt through naukri.com. http://go.msnserver.com/IN/37651.asp Post your CV online today. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * Configuration
Hi, I am a beginner to Asterisk. Can anybody clear my following doubts regarding the configuration needed? 1) What is the ideal system configuratin required?(like processer, RAM, h/d space etc) 2) How many connections it can handle at a time? 3) How many Virtual PBXs it can handle? 4) Whether Postgres or Mysql is best suited? 5) How many IVR's it can handle simultaneously? 6) How many Voicemails can be recorded at a time? 7) What type of bandwidth does * require? Thanx and Regards... Girish Gopinath _ The Great MSN Sale. Get shopping discounts. http://www.msn.co.in/Shopping Win exciting prizes! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * Configuration
From: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: Girish Gopinath [EMAIL PROTECTED] CC: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] * Configuration Date: Tue, 25 Nov 2003 15:24:11 +0500 (PKT) On Tue, 25 Nov 2003, Girish Gopinath wrote: 1) What is the ideal system configuratin required?(like processer, RAM, h/d space etc) depends (on what codec, what users, whats the purpose) 2) How many connections it can handle at a time? depends on 1 above 3) How many Virtual PBXs it can handle? depends on what a virtual pbx is expected to do, theoretical limit? 4) Whether Postgres or Mysql is best suited? depends, cdr_mysql is in addons, bkw_ just did unixodbc, what do you prefer? 5) How many IVR's it can handle simultaneously? depends on 1 6) How many Voicemails can be recorded at a time? depends (on codec, available i/o) 7) What type of bandwidth does * require? depends (on codec and protocol) Thanx and Regards... most welcome.. i was just going to write depends to all, but you get the point ... best go through the archives in detail, ALL your answers lie in that... - wasim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users _ 10,000 recruiters head-hunt through naukri.com. http://go.msnserver.com/IN/37651.asp Post your CV online today. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
