Re: [Asterisk-Users] Is this Hardaware Enough for Asterisk ?
Tarun Banka wrote: Hello, We are planning to buy following Hardware for Asterisk TestBed. Please let me know if this seems fine to you. 1. IP Phones ( 5 in number) CISCO 7940/7960, SNOM 200, Pingtel xpressa 2. Wildcard T100P interface card, that will connect Asterisk server to Our Nortel Switch SL-100 3. Wildcard TDM400P that gives us 4FXS ports for 4 Analog Phones 4. Server 1.8GHz or more P 4 1GB RAM 5. T1 Cable. Please let me know if I am missing anything. Best regards, Tarun I'm using similar setup to an old IBM PII 266/128RAM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AGI Test Fails
You're calling the script using EAGI not AGI - This caught me out the day. Changing extensions.conf to use AGI solved my problem :-) Technical explanation: Something to do with EAGI providing audio on file descriptor 3, it confuses things. Stick with using the AGI app to call your scripts and you should be fine. Cheers Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Generating a call with the Manager interface..
Hi, Currently I use call files to automate the generation of calls from our address book and the resulting call file looks like this.. Channel: SIP/201 WaitTime: 30 Application: Dial Data: CAPI/4567:5556789 CallerID: Auto Dial 1000 This method works but it not logging the calls to the CDR and there are a few other issues.. So I wanted to try and do the same thing using the manager interface in Asterisk.. The problem is that the docs are a little shy on details.. Does anyone know how I can turn my call file sample into the manager interface equivalent?? My guess is something like.. Action: Originate Channel: SIP/201 Timeout: 30 ??? (application line) ???(data line) CallerID: Auto Dial 1000 Thanks.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Error
When dialling in and dialling my extension, when answered I get Read_channel ## vpb/1-3: Setting record mode, bridge = 0 WARNING[20499]: File chan_sip.c, Line (sip_write): Asked to transmit frame type 8, while native formats is 4 (read/write = 4/4) == Spawn extension (default, 1004, 1) exited non-zero on 'vpb/1-3' -- hangup on vpb (vpb/1-3) -- Hungup on vpb/1-3 complete -- Event [12=[02] Loop Drop And it hangs up the line any ideas ??? Regards Mick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Extension Dialing problem with SIP
Hi List.. I m getting this mesg while trying to dial an extension, both SIP UAs are registered with asterisk, m trying to dial extension 1015 from UA [EMAIL PROTECTED] to extension 1016 of UA [EMAIL PROTECTED] In extensions.conf I added exten = 1015,1, Dial(SIP/7,20,tr) Any hint? JF WARNING[16397]: File pbx.c, Line 1153 (pbx_extension_helper): No application ' Dial' for extension (default, 1015, 1) == Spawn extension (default, 1015, 1) exited non-zero on 'SIP/12321-56a2'Reliably Transmitting (no NAT):SIP/2.0 403 Forbidden Do you Yahoo!? The New Yahoo! Shopping - with improved product search
Re: [Asterisk-Users] New TDM cards--driver won't load
Hi all, FYI, I had a similar problem where the new TDM card would show up in /proc/pci but would not load the module. Checked out the latest CVS zaptel, made / installed and loaded straight away. Used Gigabyte GA-8S648 board. Stuart. On Fri, 2003-10-03 at 19:49, Mark Spencer wrote: Is it showing up on /proc/pci? It should be a tigerjet. Yes. I put the other card back in (production machine) but over the weekend I'll get the card in there and capture the output of lspci. If the card shows up in /proc/pci then your motherboard *must* be supplying 3.3V somehow (unless it's just leaking back somehow, but that doesn't seem very likely). Nope. Doesn't show any sign of seeing the card at all. It does see the FXO card that's in the same machine. We should check the PCI ID's. There may be a difference in what we're looking for and what is there. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- | http://www.opusvl.com | T: 08717 50 40 02 | F: 08717 50 40 03 | E: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] replacing sound files
how do you go about replacing the sound files in * with your own ?? Regards Mick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with MeetMe.
Good afternoon, I'm trying to use MeetMe in an AGI script written in Perl, as follows: print EXEC MeetMe 2000|p \n; $res = checkresult(); The problem that I have is that when a user press '#' in order to exit from the conference, everybody goes out. This is randomized because sometimes doesn't happened. My current version of asterisk is: Asterisk CVS-05/22/03-11:14:50 built by [EMAIL PROTECTED] on a i686 running Linux Does somebody knows the problem? It's a version problem of *??? I would be very pleasured if somebody can help me or told me if I'm mistaked about this functionality of MeetMe. Thanks a lot. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bare-bone config
Hi, could somebody name the minimum configuration files asterisk needs to run with a SIP phone? what do i need apart from asterisk.conf and extensions.conf? tia -- Mit freundlichen Gren Conrad Braun Pentaprise GmbH Im Pinderpark 5 D-90513 Zirndorf http://www.pentaprise.de ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with meetme.
Good afternoon, I'm trying to use MeetMe in an AGI script written in Perl, as follows: print EXEC MeetMe 2000|p \n; $res = checkresult(); The problem that I have is that when a user press '#' in order to exit from the conference, everybody goes out. This is randomized because sometimes doesn't happened. My current version of asterisk is: Asterisk CVS-05/22/03-11:14:50 built by [EMAIL PROTECTED] on a i686 running Linux Does somebody knows the problem? It's a version problem of *??? I would be very pleasured if somebody can help me or told me if I'm mistaked about this functionality of MeetMe. Thanks a lot. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bare-bone config
Conrad Braun wrote: Hi, could somebody name the minimum configuration files asterisk needs to run with a SIP phone? what do i need apart from asterisk.conf and extensions.conf? tia Probably sip.conf.. :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] e100p in norway?
I have had a similar problem with a BT circuit, it turns out my circiut is not PRI, but DAS, which is some sort of BT enhanced (or modified) PRI, I believe the signalling is a bit different. My PRI premise gear has occasional lockups with it. note this is not an asterisk setup, but a LCR providers piece of hardware. You may just want to ensure thaqt your PRI is really PRI. James Richards, IT Manager Europe Electronic Theatre Controls -Original Message- From: Alastair Maw [mailto:[EMAIL PROTECTED] Sent: 13 October 2003 10:18 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] e100p in norway? On 13/10/03 10:03, Roy Sigurd Karlsbakk wrote: see below's conversation. it seems the e100p card doesn't work with BT. Any idea how this'll work against Telenor (norway)? roy RoyK does anyone know if I can trust the E100P to do full PRI stuff in .no? cypromis dunno about no cypromis I cannot use it in UK cypromis cause the framer has problems with system-x switches at bt I'm using an E400P in the UK with a System-X switch (provider is YourCommunications). I was previously using one with Colt (unknown switch). IIRC, getting it working required a fairly new build of libpri/*/zaptel. -- Alastair Maw MX Telecom www.mxtelecom.com ___ 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
Re: [Asterisk-Users] bare-bone config
WipeOut wrote: Conrad Braun wrote: Hi, could somebody name the minimum configuration files asterisk needs to run with a SIP phone? what do i need apart from asterisk.conf and extensions.conf? tia Probably sip.conf.. :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Mit freundlichen Grüßen Conrad Braun Pentaprise GmbH Im Pinderpark 5 D-90513 Zirndorf http://www.pentaprise.de ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bare-bone config
Conrad Braun wrote: ok, that's obvious. simply forgot to mention it ;) but do I need any of the other files at all? ps. sorry for posting an empty reply just seconds earlier... Why do you want to remove some of the conf files? Just leave them all there.. its not like they use up a lot of space or anything.. :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T100P Phones Configuration
It appears that the T1 Digium cards can split voice and data, but I would not want data traffic going through the * server... Yes, they can do this. You can turn your * server into both a PBX and router. Do you have any documentation on how to set this up? Also I was talking with James (different James, hahaha) about using a T400P to take in a PRI from the telco and provide a PRI (or CAS T1) to an old access server, routing modem calls to the access server by DID. James was unsure if 56k (v90) speeds could be achieved but his test was a little wonky too, he admitted. Do you knwo of any limitations/restrictions on doing this? Obviously modem calls coming from VOIP would have to use ulaw/alaw but being able to get v90 and voice with the T400 would be nice, if possible. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Echo Problems..What's going to happen?
I've read and experienced the echo problems with the X100P. Is Digium going to fix the problem or refund our money? I want to see this work because myself and other small companies out there use analog lines. I would trade up to T1 but that requires me to have at least 9 lines. If I did trade up, do the T1 cards work perfectly with no echo at all? I get echo with my directly connected computer using Xten SIP. No matter with all the suggestions to change the parameters, it still has echo. Can you explain what you have tried? If it's _you_ who is getting the echo (not the party calling you) and you just can't kill it, the culprit often enough seems to be your PSTN interface; near-end echo is largely caused by impedance mismatches in the hybrid circuit (the circuit which converts the 4-line interface to the 2-line one) -- the software echo cancellers just try and get rid of the echo in software but the problem is often a hardware issue. Just to reiterate the possible solutions that I am aware of: - try different echo cancel algorithms (mark2 with agressive mode is what I use) - reduce or increase the number of taps (echocancel=yes is equivalent to echocancel=128) -- the fewer taps the better, but more taps may be necessary to eliminate echo. - try reversing the ring and tip on the FXO interface - try attaching the FXO card right at the demarcation point and remove ANY other equipment connected at the point of demarcation. Stubs can play havoc with echo cancellation Does anyone have the T1 and have no problems at all? I would surely appreciate you experiences. What's my option to get this too work flawlessly? I am currently experimenting with the T1 card and a channel bank to see if telco-grade equipment will do a better job of removing echo. I'm waiting on a cable before I can start testing though. :-( I have a Carrier Access Channel Bank I with 12 FXO ports on it, and an Adit 600 with 12 FXS ports coming soon. the CBI is pretty old and I've heard that it cannot do far-end disconnect detection but the FXO cards claim that they dynamically adjust to compensate for impedance changes to minimize echo, so I'm keen on seeing what they can do. I just got pricing from our telco (Bell Canada) and to purchase an entire ISDN PRI (23B+D) with 30DID and 911 service, the per-line cost is almost identical to regular analogue PSTN ports. There'd be no echo on those babies, I imagine, since there's no hybrid circuit until you hit the other end of the call. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Processor support..
depends how large. If you are playing with the idea of setting up something like vonage but in a not so bandwidth loaded country you will need the horsepower for codec stuff. On Monday 13 October 2003 7:11 am, Chris Albertson wrote: Do you r really need more CPU power for Asterisk? I'd think in a larger system you'd go with multiple servers this would allow for redundancy --- WipeOut [EMAIL PROTECTED] wrote: Hey.. Has anyone played around with Asterisk on the Itanium2, Opteron or Athlon64?? Can Asterisk (or Linux for that matter) actually make good use of a 64bit system?? Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Bielicki Managing Director TAAN Consultants Ltd http://www.global-gateway.net/ -- This correspondence is for the named person's use only. It may contain confidential or legally privileged information or both. No confidentiality or privilege is waived or lost by any mistransmission. If you receive this correspondence in error, please immediately delete it from your system and notify the sender. You must not disclose, copy or rely on any part of this correspondence if you are not the intended recipient. Any opinions expressed in this message are those of the individual sender. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] replacing sound files
tar Regards Mick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: Monday, 13 October 2003 9:54 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] replacing sound files [EMAIL PROTECTED] wrote: how do you go about replacing the sound files in * with your own ?? There is a page on the Wiki about Asterisk sound files that describes where the files are located and what they contain. http://www.voip-info.org /O ___ 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] Gatekeeper with Asterisk
Hi! I am trying to do a SIP/H.323 gateway. I want that the SIP proxy server (I suppose that this is asterisk isn't it?) has all the information about the user's registration. So, when a request arrives at the gatekeeper from the H.323 network, this one tries to make multicast to all the others gatekeeper and also the Gatekeeper Asterisk. Asterisk then looks for if the user called is a SIP user or not. Is that possible to use Asterisk as SIP proxy and a SIP/H.323 gateway? In my network I have already a gatekeeper. If it is possible... how can I do that with asterisk? Thanks for all your help Regards, Mireia ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bare-bone config
I am just starting to use asterisk as well as VoIP in general, and it's a bit confusing finding out what goes where... in my eyes it seems to be a lot easier to start with a bare minimum, thereby eliminating as many causes for error as possible. when I feel comfortable, I can always expand on top of it. Also, I haven't found any documentation on which files are read and in what order - are the names hardcoded? why isn't there a h323.conf? so it's also a matter of curiosity I guess ;) WipeOut wrote: Why do you want to remove some of the conf files? Just leave them all there.. its not like they use up a lot of space or anything.. :) -- Mit freundlichen Grüßen Conrad Braun Pentaprise GmbH Im Pinderpark 5 D-90513 Zirndorf http://www.pentaprise.de ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T100P Phones Configuration
On Monday, October 13, 2003 7:21 AM, Andrew Kohlsmith [SMTP:[EMAIL PROTECTED] wrote: Also I was talking with James (different James, hahaha) about using a T400P to take in a PRI from the telco and provide a PRI (or CAS T1) to an old access server, routing modem calls to the access server by DID. James was unsure if 56k (v90) speeds could be achieved but his test was a little wonky too, he admitted. Do you knwo of any limitations/restrictions on doing this? Obviously modem calls coming from VOIP would have to use ulaw/alaw but being able to get v90 and voice with the T400 would be nice, if possible. 56k speeds should work fine. Going between 2 PRI will not drop any data bits. Going between a PRI and robbed bit signaling T1 will possibly change a data bit every sixth frame (.75 m seconds) but this should not noticeably effect the performance of a 56K dial up modem. (What really effects the performance is adding another A/D conversion.) Don Pobanz Regards, Andrew ___ 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
Re: [Asterisk-Users] New Processor support..
Michael Bielicki wrote: depends how large. If you are playing with the idea of setting up something like vonage but in a not so bandwidth loaded country you will need the horsepower for codec stuff. On Monday 13 October 2003 7:11 am, Chris Albertson wrote: Do you r really need more CPU power for Asterisk? I'd think in a larger system you'd go with multiple servers this would allow for redundancy --- WipeOut [EMAIL PROTECTED] wrote: Hey.. Has anyone played around with Asterisk on the Itanium2, Opteron or Athlon64?? Can Asterisk (or Linux for that matter) actually make good use of a 64bit system?? Later.. After reading a little more into it it looks like the answer to horsepower is a SMP Xeon or Athlon right now while the world slowly shifts to 64bit apps.. I guess the shift will come when M$ get up to speed.. If you really want 64bit hardware then it appears that the Opteron is preferred over the Itanium2 becasue the Opteron will suposedly run 16bit and 32bit apps with no problems seeing as AMD simply extented the x86 architecture from 32bit to 64bit where Intel tried to redesign the whole thing when they created the Itanium (Itanic as i have seen it referred to) which didn't work so well so now they have fixed some of the shortfalls in the Itaniun2.. That was my take on the articles I have read anyway.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] test calls between iaxtel fwd
Does anyone know if there is a dialplan in place that would allow me to dial out via iaxtel (with a 700 number) and back into my fwd number? I've tested fine in the opposite direction, but would like to verify the fwd incoming call success. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [Asterisk-Dev] Cisco ATA 186 chan_mgcp Transfer problem
Florian Overkamp schrieb:
Hey, if I press Flash asterisk gets the 'hf' event but does nothing.
What gives ? :-)
We can compare our ATA-configs, because transfering works fine with MGCP
(SIP doesnt).
By the way, I'd think maybe it's not actually transferring but rather
'bridging' through the ATA ?
Maybe you can show some config snippets ?
The call seems (for me) to be bridged by *:
gw-bzo*CLI
gw-bzo*CLI show channels
Channel (ContextExtensionPri ) State Appl. Data
SIP/snom1-baef (default 1 ) Up Bridged Call
CAPI[contr1/8504]/22
CAPI[contr1/8504]/22 (macro-stdexten s3 ) Up Dial
SIP/snom1|20|mt
2 active channel(s)
gw-bzo*CLI
Is this complete?
Thomas
mgcp.conf:
[general]
port = 2727
bindaddr = 0.0.0.0
disallow=all
allow=alaw
inbanddtmf=0
transfer = yes
threewaycalling=yes
musiconhold=1
[192.168.1.25]
transfer = yes
threewaycalling=yes
host = 192.168.1.25
context = default
callerid = Thomas 8504
mailbox = 8504
callgroup = 1
pickupgroup = 1
transfer=1
line = aaln/1
context = default
callerid = Thomas 8506
mailbox = 8506
transfer = 1
line = aaln/2
line = *
sip.conf:
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 192.168.1.2 ; Address to bind to
context = default ; Default for incoming calls
disallow=all
allow=ulaw
allow=alaw
allow=gsm
[snom1]
type=friend
secret=snom1
host=dynamic
defaultip=192.168.1.18
context = default
mailbox = 8501
callerid = Thomas 8501
calleridnum = 8501
callgroup = 1
pickupgroup = 1
capi.conf:
;
; CAPI config
;
;
; Multipoint
[global]
mode=immediate
isdnmode=multipoint
;nationalprefix=00
;internationalprefix=000
[interfaces]
msn=8500,8501,8503,8504,8505,8506,8507,8508,8509,8510,8511,8512,8513,8514,8515,8516,8517,8518
incomingmsn=*
controller=1,2
context=isdn
echosquelch=1
softdtmf=0
rxgain=1
txgain=1
devices=2
extensions.conf:
[general]
static=yes
writeprotect=no
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten = s,1,DBget(temp=UML/${ARG1})
exten = s,2,Goto(default|${temp}|1)
exten = s,102,Goto(s|3)
exten = s,3,Dial(${ARG2},20,mt)
exten = s,4,Voicemail2(u${ARG1})
exten = s,5,HangUp
exten = s,104,Voicemail2(b${ARG1})
exten = s,105,HangUp
[outgroup]
exten = _X.,1,Dial(CAPI/${CALLERIDNUM}:b${EXTEN},,T)
exten = _X.,2,Dial(CAPI/8501:b${EXTEN},,T)
[asterisk]
include = parkedcalls
exten = 8501,1,Macro(stdexten,8501,SIP/snom1)
exten = 8504,1,Macro(stdexten,8501,MGCP/aaln/[EMAIL PROTECTED])
[isdn]
exten = s,1,AGI(nuller.agi)
; nuller.agi adds a leading zero for incoming calls and jumps to context
; asterisk
___
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RE: [Asterisk-Users] No sound with SIP Phones on the Internet
Yes, my Asterisk is behind a NAT but I forward all ports (100-56000) to my Linux box. Chris HARIGA -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Uriel Carrasquilla Sent: Monday, October 13, 2003 12:18 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] No sound with SIP Phones on the Internet is your SIP phone behind a NAT? is* behind a NAT? Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Chris Hariga Sent: Sunday, October 12, 2003 10:42 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] No sound with SIP Phones on the Internet Hi, I need some help with my sip phones. I have a Xten softphone and a Budge Tone 101 from Grandstream. If Im connected from my LAN all is fine but from the Internet I connect the phone but I dont have the sound. Asterisk SLI show me this when I try to call my voicemail: localhost*CLI -- Executing VoiceMailMain(SIP/chariga-c067, 105) in new stack == Parsing '/etc/asterisk/voicemail.conf': == Parsing '/etc/asterisk/voicemail.conf': Found -- Playing 'vm-password' == Spawn extension (internal, 205, 1) exited non-zero on 'SIP/chariga-c067' -- Unregistered SIP 'chariga' localhost*CLI Any help is welcome. Best regards, Chris Hariga smime.p7s Description: S/MIME cryptographic signature
[Asterisk-Users] Ports open
Hi! I need to open both ports 1720 and 1719. How can I do that? Thanks. Regards, Mireia ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bare-bone config
On 13/10/03 14:05, Conrad Braun wrote: Why do you want to remove some of the conf files? Just leave them all there.. its not like they use up a lot of space or anything.. :) I am just starting to use asterisk as well as VoIP in general, and it's a bit confusing finding out what goes where... in my eyes it seems to be a lot easier to start with a bare minimum, thereby eliminating as many causes for error as possible. when I feel comfortable, I can always expand on top of it. If you just bit the bullet and removed them all, you'll discover all sorts of interesting dependencies on musiconhold, etc. On my production boxes I have autoload=no in modules.conf and then load everything in manually, as reducing the number of modules that are loaded that you don't actually use is obviously a good idea for memory footprint, stability, etc. -- Alastair Maw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bare-bone config
You will need : extensions.conf indications.conf logger.conf manager.conf rtp.conf sip.conf modules.conf ; with a crap load of stuff turned off: noload = chan_modem.so noload = chan_modem_aopen.so noload = chan_modem_bestdata.so noload = chan_modem_i4l.so noload = chan_phone.so noload = chan_mgcp.so noload = chan_iax2.so noload = chan_oss.so noload = chan_iax.so noload = chan_alsa.so noload = chan_oss.so .etc. You may want: voicemail.conf ; Do you want voicemail ? parking.conf ; Do you want to park ? meetme.conf ; Do you want a conf ? queues.conf ; Do you want queues? Conrad Braun wrote: I am just starting to use asterisk as well as VoIP in general, and it's a bit confusing finding out what goes where... in my eyes it seems to be a lot easier to start with a bare minimum, thereby eliminating as many causes for error as possible. when I feel comfortable, I can always expand on top of it. Also, I haven't found any documentation on which files are read and in what order - are the names hardcoded? why isn't there a h323.conf? so it's also a matter of curiosity I guess ;) WipeOut wrote: Why do you want to remove some of the conf files? Just leave them all there.. its not like they use up a lot of space or anything.. :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bare-bone config
Conrad Braun wrote: I am just starting to use asterisk as well as VoIP in general, and it's a bit confusing finding out what goes where... in my eyes it seems to be a lot easier to start with a bare minimum, thereby eliminating as many causes for error as possible. when I feel comfortable, I can always expand on top of it. Also, I haven't found any documentation on which files are read and in what order - are the names hardcoded? why isn't there a h323.conf? so it's also a matter of curiosity I guess ;) WipeOut wrote: Why do you want to remove some of the conf files? Just leave them all there.. its not like they use up a lot of space or anything.. :) I would say that to start you should take a read through the old and new handbooks, and then keep them handy for reference when you start playing.. The names of the .conf files are hard coded but I am sure if you were desperate enough you could edit the source code and change them.. I would suggest that you install Asterisk and start it in the default config.. then play with the .conf files that are provided to get your head around it.. take it a step at a time.. if you try to understand it all at once you are in for a hard time.. I have been using it for about 6 months now and I am still learning better or more efficient ways to do things.. The reason there is no h323.conf is becasue H.323 support is not compiled by default during installation.. you have to add it manually afterwards(and after you have met the additional pre-req's).. Good luck with your playing.. :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bare-bone config
Alastair Maw wrote: On 13/10/03 14:05, Conrad Braun wrote: Why do you want to remove some of the conf files? Just leave them all there.. its not like they use up a lot of space or anything.. :) I am just starting to use asterisk as well as VoIP in general, and it's a bit confusing finding out what goes where... in my eyes it seems to be a lot easier to start with a bare minimum, thereby eliminating as many causes for error as possible. when I feel comfortable, I can always expand on top of it. If you just bit the bullet and removed them all, you'll discover all sorts of interesting dependencies on musiconhold, etc. On my production boxes I have autoload=no in modules.conf and then load everything in manually, as reducing the number of modules that are loaded that you don't actually use is obviously a good idea for memory footprint, stability, etc. Here's a list of config files http://www.voip-info.org/wiki-Asterisk+config+files I haven't seen a barebone config, it all depends on what a barebone config is... :-) For me, it's without MGCP and for other people, how strange as it may sound, it's without SIP... /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] test calls between iaxtel fwd
Does anyone know if there is a dialplan in place that would allow me to dial out via iaxtel (with a 700 number) and back into my fwd number? I've tested fine in the opposite direction, but would like to verify the fwd incoming call success. Rich 700-99-X where X is the 5 digit FWD number. -Shaun ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] test calls between iaxtel fwd
Does anyone know if there is a dialplan in place that would allow me to dial out via iaxtel (with a 700 number) and back into my fwd number? I've tested fine in the opposite direction, but would like to verify the fwd incoming call success. Rich 700-99-X where X is the 5 digit FWD number. When I try the above, I get a party unavailable message. Since I can place outbound fwd calls, does that indicate I need to have * register with fwd (in order to accept inbound calls)? The CLI does not indicate any attempt by fwd to contact my * either. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help me please!
Hi! I have configured my gatekeeper to call asterisk everytime that the phone number begins with 064... When the gatekeeper contacts asterisk, it does it using the 1719 port, but it is closed. How can I open this port? Or the solution is to redirect the messages arriving at 1719 to 1720? Thanks for your help. Regards, Mireia ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Error
Translation: Asked to transmit frame type G.711 A-law, while native formats is G.711 u-law (read/write =G.711 u-law/G.711 u-law) Looks to me like you need a disallow=all in your sip.conf and allow= lines for the codecs you want to allow, then make sure that the IP phones you are using support at least one of the codecs you are specifying in sip.conf Try show codecs at the Asterisk CLI. On Mon, 2003-10-13 at 04:48, [EMAIL PROTECTED] wrote: When dialling in and dialling my extension, when answered I get Read_channel ## vpb/1-3: Setting record mode, bridge = 0 WARNING[20499]: File chan_sip.c, Line (sip_write): Asked to transmit frame type 8, while native formats is 4 (read/write = 4/4) == Spawn extension (default, 1004, 1) exited non-zero on 'vpb/1-3' -- hangup on vpb (vpb/1-3) -- Hungup on vpb/1-3 complete -- Event [12=[02] Loop Drop And it hangs up the line any ideas ??? Regards Mick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100P Config
Hi All. Followed the information from the link bellow and can now see the card. But. When I run modprobe zaptel I get the message that the zaptel.o was compiled for kernel version 2.4.20-4GB while this kernel version is 2.4.20-4GB-athlon. And fails. When I run modprobe wcfxo I get the message that the zaptel.o was compiled for kernel version 2.4.20-4GB while this kernel version is 2.4.20-4GB-athlon. And fails. How di I get zaptel wcfxo to recognize my kernel? I am not a linux guru so layman terms would be appreciated. Thanks in advance Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of TeleSIP Sent: 10 October 2003 23:38 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] X100P Config LeterheadThe FAQ at digium explains how to do it: http://www.digium.com/index.php?menu=faq#Configuration_7 - Original Message - From: David J Carter To: [EMAIL PROTECTED] Sent: Friday, October 10, 2003 5:29 PM Subject: RE: [Asterisk-Users] X100P Config Hi, I can see the card with a cat /proc/pci. I don't seem to have a zaptel.conf file in the etc directory. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Glenn Dalgliesh Sent: 10 October 2003 19:22 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] X100P Config What do you have configured in your /etc/zaptel.conf * /etc/asterisk/zapata.conf file. Also, submit a cat /proc/pci this should show a device Tiger Jet Network Inc. if the pci bus recognized the card. - Original Message - From: David J Carter To: [EMAIL PROTECTED] Sent: Friday, October 10, 2003 2:05 PM Subject: [Asterisk-Users] X100P Config Hiya all, I have just received my X100P telco card and I don't seem to be able to talk to it. I am a bit of a numpty on Linux being from the Windows (wash my mouth with soap and water) background, so any help would be appreciated. I have checked under YaST2 and I think it can see the card, but not sure. My * box is talking between 2 Grandstream phones no probs but now I would like to talk to the outside world. Thanks in anticipation. Dave ___ 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
Re: [Asterisk-Users] Extension Dialing problem with SIP
Remove the space before Dial On Mon, 2003-10-13 at 05:27, John Foster wrote: Hi List.. I m getting this mesg while trying to dial an extension, both SIP UAs are registered with asterisk, m trying to dial extension 1015 from UA [EMAIL PROTECTED] to extension 1016 of UA [EMAIL PROTECTED] In extensions.conf I added exten = 1015,1, Dial(SIP/7,20,tr) Any hint? JF WARNING[16397]: File pbx.c, Line 1153 (pbx_extension_helper): No application ' Dial' for extension (default, 1015, 1) == Spawn extension (default, 1015, 1) exited non-zero on 'SIP/12321-56a2' Reliably Transmitting (no NAT): SIP/2.0 403 Forbidden __ Do you Yahoo!? The New Yahoo! Shopping - with improved product search __ This message has been 'sanitized'. This means that potentially dangerous content has been rewritten or removed. The following log describes which actions were taken. Sanitizer (start=1066041054): Part (pos=2382): SanitizeFile (filename=unnamed.txt, mimetype=text/plain): Match (names=unnamed.txt, rule=1): ScanFile (file=/tmp/att-3f8a7ede-SLO-unnamed.txt): Scan succeeded, file is clean. Enforced policy: unknown Match (names=unnamed.txt, rule=3): Enforced policy: accept Part (pos=3152): SanitizeFile (filename=unnamed.html, mimetype=text/html): Match (names=unnamed.html, rule=1): ScanFile (file=/tmp/att-3f8a7ee0-6ZP-unnamed.html): Scan succeeded, file is clean. Enforced policy: unknown Match (names=unnamed.html, rule=3): Enforced policy: accept Note: Styles and layers give attackers many tools to fool the user and common browsers interpret Javascript code found within style definitions. References: - http://www.securityfocus.com/bid/630 - http://archives.indenial.com/hypermail/bugtraq/2001/January2001/0512.html Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Total modifications so far: 28 Anomy 0.0.0 : Sanitizer.pm $Id: Sanitizer.pm,v 1.79 2003/06/19 19:22:00 bre Exp $ -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bare-bone config
Try http://www.fnords.org/~eric/asterisk/ It contains simplified config files as well as other information. On Mon, 2003-10-13 at 06:34, Conrad Braun wrote: Hi, could somebody name the minimum configuration files asterisk needs to run with a SIP phone? what do i need apart from asterisk.conf and extensions.conf? tia -- Mit freundlichen Gren Conrad Braun Pentaprise GmbH Im Pinderpark 5 D-90513 Zirndorf http://www.pentaprise.de ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Strange message !! Unknown IE 76 (Unknown Information Element)
Here is an example call (works) : -- Executing Dial(SIP/25-e804, Zap/g1/0707038340) in new stack -- Called g1/0707038340 -- Zap/1-1 is ringing !! Unknown IE 76 (Unknown Information Element) -- Zap/1-1 answered SIP/25-e804 What does that !! Unknown IE 76 (Unknown Information Element) mean ?? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Gates steps up telecom campaign
Will M$ ever stop!!.. Whats the bet their telecoms products will use non-standard protocols.. I really wouldn't like to run a telecom system on Windoze in the first place.. Full Story.. http://news.zdnet.co.uk/communications/0,39020336,39117099,00.htm ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange message !! Unknown IE 76 (Unknown Information Element)
Hi Marcel, IE 76 is COLP (Connected Line ID Presentation). Your telco is so kind to tell you to which number your calls has been connected. Noting to worry about... regards kapejod -- Klaus-Peter Junghanns CEO,CTO Junghanns.NET GmbH Breite Strasse 13 - 12167 Berlin - Germany fon:+49 30 79705392 fax:+49 30 79705391 iaxtel: 1-700-157-8753 email: [EMAIL PROTECTED] http://www.junghanns.net/asterisk Am Mon, 2003-10-13 um 16.56 schrieb Marcel Prisi: Here is an example call (works) : -- Executing Dial(SIP/25-e804, Zap/g1/0707038340) in new stack -- Called g1/0707038340 -- Zap/1-1 is ringing !! Unknown IE 76 (Unknown Information Element) -- Zap/1-1 answered SIP/25-e804 What does that !! Unknown IE 76 (Unknown Information Element) mean ?? Thanks ___ 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
Re: [Asterisk-Users] Strange message !! Unknown IE 76 (Unknown Information Element)
It means that this IE is not implemented in the libpri or is not very standarized. regards Martin On Mon, 13 Oct 2003, Marcel Prisi wrote: Here is an example call (works) : -- Executing Dial(SIP/25-e804, Zap/g1/0707038340) in new stack -- Called g1/0707038340 -- Zap/1-1 is ringing !! Unknown IE 76 (Unknown Information Element) -- Zap/1-1 answered SIP/25-e804 What does that !! Unknown IE 76 (Unknown Information Element) mean ?? Thanks ___ 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
Re: [Asterisk-Users] Extension Dialing problem with SIP
put a comma after "Dial" - Original Message - From: John Foster To: [EMAIL PROTECTED] Sent: Monday, October 13, 2003 5:27 AM Subject: [Asterisk-Users] Extension Dialing problem with SIP Hi List.. I m getting this mesg while trying to dial an extension, both SIP UAs are registered with asterisk, m trying to dial extension 1015 from UA [EMAIL PROTECTED] to extension 1016 of UA [EMAIL PROTECTED] In extensions.conf I added exten = 1015,1, Dial(SIP/7,20,tr) Any hint? JF WARNING[16397]: File pbx.c, Line 1153 (pbx_extension_helper): No application ' Dial' for extension (default, 1015, 1) == Spawn extension (default, 1015, 1) exited non-zero on 'SIP/12321-56a2'Reliably Transmitting (no NAT):SIP/2.0 403 Forbidden Do you Yahoo!?The New Yahoo! Shopping - with improved product search
RE: [Asterisk-Users] X100P Config
Thanks Rich, I am re-installing the base SuSE Linux system again and will try to install everything without doing any updates. I can't remember any updates being done, but these automated installs for numpties like me could do anything and I wouldn't know. I will let you know how it goes. Cheers Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rich Adamson Sent: 13 October 2003 17:12 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] X100P Config When I run modprobe zaptel I get the message that the zaptel.o was compiled for kernel version 2.4.20-4GB while this kernel version is 2.4.20-4GB-athlon. And fails. When I run modprobe wcfxo I get the message that the zaptel.o was compiled for kernel version 2.4.20-4GB while this kernel version is 2.4.20-4GB-athlon. And fails. That's a real common problem discussed several times in the list. The issue is that somewhere along the line you've upgraded the kernel binaries (probably RedHat's up2date), and the source code that was installed in your base system (probably header files only) are from an earlier kernel. You'll need to install the kernel source for the actual version you are running. ___ 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
Re: [Asterisk-Users] Strange message !! Unknown IE 76 (Unknown Information Element)
Hi Martin, it's not implemented in libpri but very well standarized (ETS 300 097). regards, kapejod -- Klaus-Peter Junghanns CEO,CTO Junghanns.NET GmbH Breite Strasse 13 - 12167 Berlin - Germany fon:+49 30 79705392 fax:+49 30 79705391 iaxtel: 1-700-157-8753 email: [EMAIL PROTECTED] http://www.junghanns.net/asterisk Am Mon, 2003-10-13 um 17.24 schrieb Martin Pycko: It means that this IE is not implemented in the libpri or is not very standarized. regards Martin On Mon, 13 Oct 2003, Marcel Prisi wrote: Here is an example call (works) : -- Executing Dial(SIP/25-e804, Zap/g1/0707038340) in new stack -- Called g1/0707038340 -- Zap/1-1 is ringing !! Unknown IE 76 (Unknown Information Element) -- Zap/1-1 answered SIP/25-e804 What does that !! Unknown IE 76 (Unknown Information Element) mean ?? Thanks ___ 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
Re: [Asterisk-Users] Strange message !! Unknown IE 76 (Unknown Information Element)
My fault then :) I was thinking only in terms of Q931 spec ... Martin On 13 Oct 2003, Klaus-Peter Junghanns wrote: Hi Martin, it's not implemented in libpri but very well standarized (ETS 300 097). regards, kapejod -- Klaus-Peter Junghanns CEO,CTO Junghanns.NET GmbH Breite Strasse 13 - 12167 Berlin - Germany fon: +49 30 79705392 fax: +49 30 79705391 iaxtel: 1-700-157-8753 email:[EMAIL PROTECTED] http://www.junghanns.net/asterisk Am Mon, 2003-10-13 um 17.24 schrieb Martin Pycko: It means that this IE is not implemented in the libpri or is not very standarized. regards Martin On Mon, 13 Oct 2003, Marcel Prisi wrote: Here is an example call (works) : -- Executing Dial(SIP/25-e804, Zap/g1/0707038340) in new stack -- Called g1/0707038340 -- Zap/1-1 is ringing !! Unknown IE 76 (Unknown Information Element) -- Zap/1-1 answered SIP/25-e804 What does that !! Unknown IE 76 (Unknown Information Element) mean ?? Thanks ___ 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No sound with SIP Phones on the Internet
This is bull... I can't believe that... Must be a solution... Chris HARIGA -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of WipeOut Sent: Monday, October 13, 2003 9:57 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] No sound with SIP Phones on the Internet Chris Hariga wrote: Yes, my Asterisk is behind a NAT but I forward all ports (100-56000) to my Linux box. There is your problem.. Asterisk does not like playing behind NAT.. The UA's can be made to work behind NAT but the server must have a public IP address.. ___ 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
RE: [Asterisk-Users] No sound with SIP Phones on the Internet
This is bull... I can't believe that... Must be a solution... sip is very tricky to get working behind firewalls. sip clients work quite well with nat, just make sure nat=yes is in the sip profile in sip.conf my solution has always been to put an asterisk box behind the firewall and make all the sip clients connect to that, then IAX out of the firewall to the other machines. i spent a few days trying unsuccessfully to find a decent sip proxy that worked the way i wanted and decided that the asterisk solution was much better. duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No sound with SIP Phones on the Internet
Chris Hariga wrote: This is bull... I can't believe that... Must be a solution... Chris HARIGA -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of WipeOut Sent: Monday, October 13, 2003 9:57 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] No sound with SIP Phones on the Internet Chris Hariga wrote: Yes, my Asterisk is behind a NAT but I forward all ports (100-56000) to my Linux box. There is your problem.. Asterisk does not like playing behind NAT.. The UA's can be made to work behind NAT but the server must have a public IP address.. There is a solution.. buy a SIP aware router with a built in SIP proxy.. But even then you will probably still have issues.. Search the archives and you will see that this issue has come up time and time again and I have not heard of anyone who has managed to get Asterisk to work correctly when the Asterisk server is behind NAT.. If the SIP UA is also behind NAT then there is even less chance of it working.. Believe it, Don't believe it its your choice.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange message !! Unknown IE 76 (Unknown Information Element)
Klaus-Peter Junghanns wrote: Hi Martin, libpri misses all the fun stuff :-( hold, retrieve, suspend, ect, cd, conf, 3pty .. but i am going to change that :-) regards kapejod It misses all the timers, too. :-) Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323 - Segmentation fault (core dumped)
Hi all:
I've got some core dumps when I use chan_h323. I dial an extension using
h323, routed thru an E100P (like a H323-ISDN_PRI gateway). Sometimes *
hangs, sometimes not. The client used for test es SjPhone
(http://www.sjlabs.com/).
This is the data for one core dump:
(gdb) bt
#0 ast_rtp_get_us (rtp=0x0, us=0x5074759c) at rtp.c:790
#1 0x41f8879c in create_connection (call_reference=1349809548) at
chan_h323.c:928
#2 0x41f8f34b in
MyH323Connection::CreateRealTimeLogicalChannel(H323Capability const,
H323Channel::Directions, unsigned, H245_H2250LogicalChannelParameters
const*) (this=0x8178758, [EMAIL PROTECTED], dir=IsTransmitter,
sessionID=1)
at ast_h323.cpp:626
#3 0x49470170 in H323RealTimeCapability::CreateChannel(H323Connection,
H323Channel::Directions, unsigned, H245_H2250LogicalChannelParameters
const*) const () from /root/openh323/lib/libh323_linux_x86_r.so.1
#4 0x4946071d in H245NegLogicalChannel::OpenWhileLocked(H323Capability
const, unsigned, unsigned) ()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#5 0x494604e6 in H245NegLogicalChannel::Open(H323Capability const,
unsigned, unsigned) ()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#6 0x49462423 in H245NegLogicalChannels::Open(H323Capability const,
unsigned, unsigned) ()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#7 0x4944d311 in H323Connection::OpenLogicalChannel(H323Capability const,
unsigned, H323Channel::Directions) ()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#8 0x4944cf98 in H323Connection::SelectDefaultLogicalChannel(unsigned) ()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#9 0x4944c9d2 in H323Connection::OnSelectLogicalChannels() () from
/root/openh323/lib/libh323_linux_x86_r.so.1
#10 0x4944c8b1 in H323Connection::InternalEstablishedConnectionCheck() ()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#11 0x4944a6d1 in H323Connection::HandleControlData(PPER_Stream) () from
/root/openh323/lib/libh323_linux_x86_r.so.1
#12 0x4944a28c in H323Connection::HandleControlChannel() () from
/root/openh323/lib/libh323_linux_x86_r.so.1
#13 0x494992ee in H245TransportThread::Main() () from
/root/openh323/lib/libh323_linux_x86_r.so.1
#14 0x48d33177 in PThread::PX_ThreadStart(void*) () from
/root/pwlib/lib/libpt_linux_x86_r.so.1
#15 0x4003b2b6 in start_thread () from /lib/tls/libpthread.so.0
And this is the console log:
== New H.323 Connection created.
-- Received SETUP message...
== Setting up Call
-- Calling party name: [Gustavo]
-- Calling party number: [1152880056]
-- Called party name: [0111553037260]
-- Called party number: [0111553037260]
e164: [0111553037263]
-- Executing Dial(H323/ip$10.60.144.14:1240/4096,
Zap/1/0111553037260) in new stack
-- Called 1/0111553037260
-- Channel 1, span 1 got hangup
-- Hungup 'Zap/1-1'
== No one is available to answer at this time
=*= In CreateRealTimeLogicalChannel for call 4096
-- externalIpAddress: 172.16.254.107
-- externalPort: 13488
-- SessionID: 1
-- Direction: IsTransmitter
-- Started logical channel: sending G.711-ALaw-64k{sw}
-- channelsOpen = 2
-- remoteIpAddress: 0.0.0.0
-- remotePort: 0
-- ExternalIpAddress: 172.16.254.107
-- ExternalPort: 13488
-- Gustavo has stopped calling
== H.323 Connection deleted.
-- Gustavo has stopped calling
== H.323 Connection deleted.
-- Call with ended abnormally
== H.323 Connection deleted.
channelsOpen = 1
-- Closing logical channel...
channelsOpen = 0
Segmentation fault (core dumped)
[EMAIL PROTECTED] asterisk]#
What is wrong?
Thanks in advance,
Gus
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[Asterisk-Users] PRI/E1: machine freeze/dies after a few calls
Hi all, inside my * is a E400P. The machine is a PII 400Mhz with 256MB Ram. OS is Debian woody. * is the newest cvs co. I have written a little callgen script which make outgoing calls through my *: #! /bin/sh set -e n=$1# Nummer anz=$2 # Anzhal der Versuche anz2=$3 # Kanäle sle=$4 # Timeout bis zum nächsten Versuch if [ -z $4 ]; then sle=0 fi s=1 i=1 while [ $s -le $anz ];do echo $s try started... while [ $i -le $anz2 ];do echo -e Channel: Zap/g1/$n$i\nMaxRetries: 0\nContext: callgen\nExtension: 1\nPriority: 1\nCallerid:334778\n /var/spool/asterisk/outgoing/call.$i.$s sleep 2 i=$((i + 1)) done i=1 echo sleep for $sle sec. sleep $sle s=$((s + 1)) done The calls goes out over the first two ports and through a pri switch (teles) they come back at the other two ports (3 and 4). But after a few calls my machine is completly freezed! So that i have to restart my machine. Here're my extension.conf, zapata.conf and zaptel.conf: extension.conf: [pri1] exten = _X.,1,SetAccount(pritest) exten = _X.,2,Answer exten = _X.,3,Wait(15) exten = _X.,4,Hangup [pri2] exten = _X.,1,SetAccount(pritest) exten = _X.,2,Answer exten = _X.,3,Wait(15) exten = _X.,4,Hangup [pri3] exten = _X.,1,SetAccount(pritest) exten = _X.,2,Answer exten = _X.,3,Wait(60) exten = _X.,4,Hangup [pri4] exten = _X.,1,SetAccount(pritest) exten = _X.,2,Answer exten = _X.,3,Wait(60) exten = _X.,4,Hangup [callgen] exten = 1,1,Wait(90) zapata.conf: ; ; Zapata telephony interface ; ; Configuration file [channels] pridialplan=local switchtype=euroisdn busydetect=yes callprogress=no echocancel=yes echocancelwhenbridged=yes ;callwaitingcallerid=no ;callwaiting=no signalling=pri_net group=1 context=pri1 channel = 1-15,17-31 channel =32-46,48-62 signalling=pri_net group=3 context=pri3 channel = 63-77,79-93 channel = 94-108,110-124 zaptel.conf span=1,0,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 span=3,0,0,ccs,hdb3,crc4 span=4,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 bchan=63-77,79-93 dchan=78 bchan=94-108,110-124 dchan=109 loadzone = fr defaultzone=fr Thanks for your help. Regards, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 - Segmentation fault (core dumped)
Are you using the recommended pwlib and openh323 tarballs?
bkw
On Mon, 13 Oct 2003, CW_ASN wrote:
Hi all:
I've got some core dumps when I use chan_h323. I dial an extension using
h323, routed thru an E100P (like a H323-ISDN_PRI gateway). Sometimes *
hangs, sometimes not. The client used for test es SjPhone
(http://www.sjlabs.com/).
This is the data for one core dump:
(gdb) bt
#0 ast_rtp_get_us (rtp=0x0, us=0x5074759c) at rtp.c:790
#1 0x41f8879c in create_connection (call_reference=1349809548) at
chan_h323.c:928
#2 0x41f8f34b in
MyH323Connection::CreateRealTimeLogicalChannel(H323Capability const,
H323Channel::Directions, unsigned, H245_H2250LogicalChannelParameters
const*) (this=0x8178758, [EMAIL PROTECTED], dir=IsTransmitter,
sessionID=1)
at ast_h323.cpp:626
#3 0x49470170 in H323RealTimeCapability::CreateChannel(H323Connection,
H323Channel::Directions, unsigned, H245_H2250LogicalChannelParameters
const*) const () from /root/openh323/lib/libh323_linux_x86_r.so.1
#4 0x4946071d in H245NegLogicalChannel::OpenWhileLocked(H323Capability
const, unsigned, unsigned) ()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#5 0x494604e6 in H245NegLogicalChannel::Open(H323Capability const,
unsigned, unsigned) ()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#6 0x49462423 in H245NegLogicalChannels::Open(H323Capability const,
unsigned, unsigned) ()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#7 0x4944d311 in H323Connection::OpenLogicalChannel(H323Capability const,
unsigned, H323Channel::Directions) ()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#8 0x4944cf98 in H323Connection::SelectDefaultLogicalChannel(unsigned) ()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#9 0x4944c9d2 in H323Connection::OnSelectLogicalChannels() () from
/root/openh323/lib/libh323_linux_x86_r.so.1
#10 0x4944c8b1 in H323Connection::InternalEstablishedConnectionCheck() ()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#11 0x4944a6d1 in H323Connection::HandleControlData(PPER_Stream) () from
/root/openh323/lib/libh323_linux_x86_r.so.1
#12 0x4944a28c in H323Connection::HandleControlChannel() () from
/root/openh323/lib/libh323_linux_x86_r.so.1
#13 0x494992ee in H245TransportThread::Main() () from
/root/openh323/lib/libh323_linux_x86_r.so.1
#14 0x48d33177 in PThread::PX_ThreadStart(void*) () from
/root/pwlib/lib/libpt_linux_x86_r.so.1
#15 0x4003b2b6 in start_thread () from /lib/tls/libpthread.so.0
And this is the console log:
== New H.323 Connection created.
-- Received SETUP message...
== Setting up Call
-- Calling party name: [Gustavo]
-- Calling party number: [1152880056]
-- Called party name: [0111553037260]
-- Called party number: [0111553037260]
e164: [0111553037263]
-- Executing Dial(H323/ip$10.60.144.14:1240/4096,
Zap/1/0111553037260) in new stack
-- Called 1/0111553037260
-- Channel 1, span 1 got hangup
-- Hungup 'Zap/1-1'
== No one is available to answer at this time
=*= In CreateRealTimeLogicalChannel for call 4096
-- externalIpAddress: 172.16.254.107
-- externalPort: 13488
-- SessionID: 1
-- Direction: IsTransmitter
-- Started logical channel: sending G.711-ALaw-64k{sw}
-- channelsOpen = 2
-- remoteIpAddress: 0.0.0.0
-- remotePort: 0
-- ExternalIpAddress: 172.16.254.107
-- ExternalPort: 13488
-- Gustavo has stopped calling
== H.323 Connection deleted.
-- Gustavo has stopped calling
== H.323 Connection deleted.
-- Call with ended abnormally
== H.323 Connection deleted.
channelsOpen = 1
-- Closing logical channel...
channelsOpen = 0
Segmentation fault (core dumped)
[EMAIL PROTECTED] asterisk]#
What is wrong?
Thanks in advance,
Gus
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[Asterisk-Users] PrePaid Application!!!!!
Hello, Here in our office we are testing Asterisk. My collage Igor created to Asterisk PrePaid applicationwith Postgresql. It is not in Perl. We would like to release it to the group as soon as it will work ok. It will have authentication, different rates for users, different rates for destinations and so on. Is there anybody who would like to improve it ? -- Bart
RE: [Asterisk-Users] PrePaid Application!!!!!
Title: Message I'd like to see it. What language is it in? I'm sure everyone in the group could benefit in some form --James Cornman [EMAIL PROTECTED]Completely Reliable Network Conceptshttp://www.crnc.net(v) 973-784-0031(f) 973-784-0038 -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz JozwiakSent: Monday, October 13, 2003 3:49 PMTo: ASTERISK USERSSubject: [Asterisk-Users] PrePaid Application! Hello, Here in our office we are testing Asterisk. My collage Igor created to Asterisk PrePaid applicationwith Postgresql. It is not in Perl. We would like to release it to the group as soon as it will work ok. It will have authentication, different rates for users, different rates for destinations and so on. Is there anybody who would like to improve it ? -- Bart
RE: [Asterisk-Users] PrePaid Application!!!!!
In what language is it written in? It would be interesting to at least look at it and maybe convert it to use MySQL instead -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Bartosz Jozwiak Sent: Monday, October 13, 2003 3:49 PM To: ASTERISK USERS Subject: [Asterisk-Users] PrePaid Application! Hello, Here in our office we are testing Asterisk. My collage Igor created to Asterisk PrePaid applicationwith Postgresql. It is not in Perl. We would like to release it to the group as soon as it will work ok. It will have authentication, different rates for users, different rates for destinations and so on. Is there anybody who would like to improve it ? -- Bart
Re: [Asterisk-Users] IAXTEL/ Dial problem
Hello I am still having problems with IAXTELL and FWD configuration. I get the following when I dial 17009965342 which is set as an example to dial to FWD people. 1+700+99+ 5 digit number. I have placed X where my passwords are. my IAX.conf has register = abatista:[EMAIL PROTECTED]/114 I also have FWD setup as register = 65342:[EMAIL PROTECTED]/114 So what am I doing wrong? Here's what works for me Move the register command for fwd to the sip.conf file (towards the top) as fwd apparently interfaces using SIP (not iax). Then remove the /114 from that statement. Create a sip.conf context similar to: [fwd] ; handles FWD SIP (not IAX) calls type=friend host=fwd.pulver.com username=65342 secret=XX context=fromfwd nat=no reinvite=no canreinvite=no Then in extensions.conf, something like: [fromfwd] exten = s,1,Dial(SIP/3000,20,tr) exten = s,2,VoiceMail,u3000 exten = s,102,VoiceMail,b3000 Personally, I'd remove the /114 from the iaxtel register statement as well until you have a working config. Then experiment with optional parameters. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXTEL/ Dial problem
From: Brian West [EMAIL PROTECTED]
register = abatista:[EMAIL PROTECTED]/114 doesn't work in iax.conf
also you are sending the full 917009965342
you should only send ${EXTEN:1} strip that 9 off.
OK done I forgot about the stripping the 9 off. Now I can call the numbers, But now
how do I get the call into my system? How to I route it to an extension.
bkw
On Mon, 13 Oct 2003, Ariel Batista wrote:
Hello I am still having problems with IAXTELL and FWD configuration. I get the
following when I dial 17009965342 which is set as an example to dial to FWD people.
1+700+99+ 5 digit number. I have placed X where my passwords are.
CLI Executing Dial(Zap/14-1, IAX/abatista:[EMAIL PROTECTED]/[EMAIL
PROTECTED]) in new stack
-- Calling using options 'exten=917009965342;callerid=Ariel
Batista114;language=en;context=iaxtel;username=abatista;formats=4;capability=2147483518;version=1;adsicpe=2'
-- Called abatista:[EMAIL PROTECTED]/[EMAIL PROTECTED]
-- Hungup 'IAX[12.37.165.130:5036]/7'
== No one is available to answer at this time
my IAX.conf has
register = abatista:[EMAIL PROTECTED]/114
I also have FWD setup as
register = 65342:[EMAIL PROTECTED]/114
So what am I doing wrong? I have read and done just about all the different
examples on the google search. But I am still at a lost! I have xten configure to
get fwd calls and it works. But I would like to be able to get them through my
Asterisk server to my extensions.
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[Asterisk-Users] ACD/IVR dialogs/SIP/client environment
Ok I have tried to post to this list server but have just gotten the automated reply saying the moderator has to approve it to the list first which was my mistake for sending from the wrong email account. So if the moderator finally approves my questions and you see the same post again Sorry. My situation is this: I havn't installed Asterisk yet but am curious the general way you would go about doing an ACD to A SIP client and causing an application to pop on the client side. I had thought that the way to do that would be to trigger two different events from inside an IVR on the Asterisk server. Is that possible? Can you send the call into a dialog/code that will determine the client via the DID/DNIS and then call and pass variable to an application that will communicate over the network with the clients app and then load a web page. Then the next part of the dialog/code would initiate the SIP session with the clients station? Any answers or replies to help distill this question would help. Thanks.
Re: [Asterisk-Users] PrePaid Application!!!!!
Bartosz Jozwiak wrote: Hello, Here in our office we are testing Asterisk. My collage Igor created to Asterisk PrePaid application with Postgresql. It is not in Perl. We would like to release it to the group as soon as it will work ok. It will have authentication, different rates for users, different rates for destinations and so on. Is there anybody who would like to improve it ? -- Bart I also wouldn't mind taking a look.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PrePaid Application!!!!!
UnixODBC. No need to rewrite everything for a simple DB change. In what language is it written in? It would be interesting to at least look at it and maybe convert it to use MySQL instead. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz Jozwiak Sent: Monday, October 13, 2003 3:49 PM To: ASTERISK USERS Subject: [Asterisk-Users] PrePaid Application! Hello, Here in our office we are testing Asterisk. My collage Igor created to Asterisk PrePaid application with Postgresql. It is not in Perl. We would like to release it to the group as soon as it will work ok. It will have authentication, different rates for users, different rates for destinations and so on. Is there anybody who would like to improve it ? -- Bart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXTEL/ Dial problem
-- Original Message -- From: Rich Adamson [EMAIL PROTECTED] OK I got it working. Thank you Rich I used your examples and had to add the following the sip.conf file. It did not work until I had the :5060 on it! register = 65342:[EMAIL PROTECTED]:5060 Here's what works for me Move the register command for fwd to the sip.conf file (towards the top) as fwd apparently interfaces using SIP (not iax). Then remove the /114 from that statement. Create a sip.conf context similar to: [fwd] ; handles FWD SIP (not IAX) calls type=friend host=fwd.pulver.com username=65342 secret=XX context=fromfwd nat=no reinvite=no canreinvite=no Then in extensions.conf, something like: [fromfwd] exten = s,1,Dial(SIP/3000,20,tr) exten = s,2,VoiceMail,u3000 exten = s,102,VoiceMail,b3000 Personally, I'd remove the /114 from the iaxtel register statement as well until you have a working config. Then experiment with optional parameters. ___ 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
Re: [Asterisk-Users] chan_h323 - Segmentation fault (core dumped)
Yes, I donwload tgz's from nufone (http://www.nufone.net/downloads/). All
sources was compiled as Jeremy recommeds, and I didn't have troubles with
that. Oh, I'm using RH9.
This is my h323.conf:
[general]
port = 1720
bindaddr = 0.0.0.0
disallow=all
allow=alaw
allow=gsm
dtmfmode=rfc2833
gatekeeper = DISABLE
[Gustavo]
type=user
host=10.60.144.14
context=default
incominglimit=31
Regards,
Gus
- Original Message -
From: Brian West [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, October 13, 2003 4:46 PM
Subject: Re: [Asterisk-Users] chan_h323 - Segmentation fault (core dumped)
Are you using the recommended pwlib and openh323 tarballs?
bkw
On Mon, 13 Oct 2003, CW_ASN wrote:
Hi all:
I've got some core dumps when I use chan_h323. I dial an extension using
h323, routed thru an E100P (like a H323-ISDN_PRI gateway). Sometimes *
hangs, sometimes not. The client used for test es SjPhone
(http://www.sjlabs.com/).
This is the data for one core dump:
(gdb) bt
#0 ast_rtp_get_us (rtp=0x0, us=0x5074759c) at rtp.c:790
#1 0x41f8879c in create_connection (call_reference=1349809548) at
chan_h323.c:928
#2 0x41f8f34b in
MyH323Connection::CreateRealTimeLogicalChannel(H323Capability const,
H323Channel::Directions, unsigned, H245_H2250LogicalChannelParameters
const*) (this=0x8178758, [EMAIL PROTECTED], dir=IsTransmitter,
sessionID=1)
at ast_h323.cpp:626
#3 0x49470170 in H323RealTimeCapability::CreateChannel(H323Connection,
H323Channel::Directions, unsigned, H245_H2250LogicalChannelParameters
const*) const () from /root/openh323/lib/libh323_linux_x86_r.so.1
#4 0x4946071d in H245NegLogicalChannel::OpenWhileLocked(H323Capability
const, unsigned, unsigned) ()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#5 0x494604e6 in H245NegLogicalChannel::Open(H323Capability const,
unsigned, unsigned) ()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#6 0x49462423 in H245NegLogicalChannels::Open(H323Capability const,
unsigned, unsigned) ()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#7 0x4944d311 in H323Connection::OpenLogicalChannel(H323Capability
const,
unsigned, H323Channel::Directions) ()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#8 0x4944cf98 in H323Connection::SelectDefaultLogicalChannel(unsigned)
()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#9 0x4944c9d2 in H323Connection::OnSelectLogicalChannels() () from
/root/openh323/lib/libh323_linux_x86_r.so.1
#10 0x4944c8b1 in H323Connection::InternalEstablishedConnectionCheck()
()
from /root/openh323/lib/libh323_linux_x86_r.so.1
#11 0x4944a6d1 in H323Connection::HandleControlData(PPER_Stream) ()
from
/root/openh323/lib/libh323_linux_x86_r.so.1
#12 0x4944a28c in H323Connection::HandleControlChannel() () from
/root/openh323/lib/libh323_linux_x86_r.so.1
#13 0x494992ee in H245TransportThread::Main() () from
/root/openh323/lib/libh323_linux_x86_r.so.1
#14 0x48d33177 in PThread::PX_ThreadStart(void*) () from
/root/pwlib/lib/libpt_linux_x86_r.so.1
#15 0x4003b2b6 in start_thread () from /lib/tls/libpthread.so.0
And this is the console log:
== New H.323 Connection created.
-- Received SETUP message...
== Setting up Call
-- Calling party name: [Gustavo]
-- Calling party number: [1152880056]
-- Called party name: [0111553037260]
-- Called party number: [0111553037260]
e164: [0111553037263]
-- Executing Dial(H323/ip$10.60.144.14:1240/4096,
Zap/1/0111553037260) in new stack
-- Called 1/0111553037260
-- Channel 1, span 1 got hangup
-- Hungup 'Zap/1-1'
== No one is available to answer at this time
=*= In CreateRealTimeLogicalChannel for call 4096
-- externalIpAddress: 172.16.254.107
-- externalPort: 13488
-- SessionID: 1
-- Direction: IsTransmitter
-- Started logical channel: sending G.711-ALaw-64k{sw}
-- channelsOpen = 2
-- remoteIpAddress: 0.0.0.0
-- remotePort: 0
-- ExternalIpAddress: 172.16.254.107
-- ExternalPort: 13488
-- Gustavo has stopped calling
== H.323 Connection deleted.
-- Gustavo has stopped calling
== H.323 Connection deleted.
-- Call with ended abnormally
== H.323 Connection deleted.
channelsOpen = 1
-- Closing logical channel...
channelsOpen = 0
Segmentation fault (core dumped)
[EMAIL PROTECTED] asterisk]#
What is wrong?
Thanks in advance,
Gus
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[Asterisk-Users] Call Parking and Paid Digium software modifications
Hello, I'm considering paying Digium to do a modification to Asterisk so that calls can be parked on specific user-defined numbers(transfer to 701 and it's parked on 701, transfer to 702 and it's parked on 702) instead of the way Asterisk currently does call parking(transfer to 700 and then it tells you where it put the call 701-720). What would be the price range for this feature to be programmed? Would anyone else out there be willing to contribute money for this project? Is there any chance of this being done by someone for free in the next month? Could we also add call logging to the call_parking application(for CDR and h flag)? I would, of course, want all of the code to be Open-sourced and included in Asterisk distros if possible. Any feedback would be appreciated. MATT--- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PrePaid Application!!!!!
It has been written in the same language as other applications like Dial,Queue,Record and so on. I hope that our company will say YES say we can release it. -- Bart - Bartosz Jozwiak wrote: Hello, Here in our office we are testing Asterisk. My collage Igor created to Asterisk PrePaid application with Postgresql. It is not in Perl. We would like to release it to the group as soon as it will work ok. It will have authentication, different rates for users, different rates for destinations and so on. Is there anybody who would like to improve it ? -- Bart I also wouldn't mind taking a look.. Later.. - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Parking and Paid Digium software modifications
Is there an underlying reason you want to do this? Because if a call is already parked on 701 and you transfer another call to 701 to park it, both callers would be connected. I am sure there is a better way to implement what you want. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of mattf Sent: Monday, October 13, 2003 5:44 PM To: '[EMAIL PROTECTED]' Subject: [Asterisk-Users] Call Parking and Paid Digium software modifications Hello, I'm considering paying Digium to do a modification to Asterisk so that calls can be parked on specific user-defined numbers(transfer to 701 and it's parked on 701, transfer to 702 and it's parked on 702) instead of the way Asterisk currently does call parking(transfer to 700 and then it tells you where it put the call 701-720). What would be the price range for this feature to be programmed? Would anyone else out there be willing to contribute money for this project? Is there any chance of this being done by someone for free in the next month? Could we also add call logging to the call_parking application(for CDR and h flag)? I would, of course, want all of the code to be Open-sourced and included in Asterisk distros if possible. Any feedback would be appreciated. MATT--- ___ 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
Re: [Asterisk-Users] Gates steps up telecom campaign
On Mon, 13 Oct 2003 14:01:00 -0400, Andrew Kohlsmith wrote: I really wouldn't like to run a telecom system on Windoze in the first place.. One of the Meridian agent systems uses OS/2 on their system... :-) mmm, thanks for reminding me, i still have one system running OS/2. I hadn't looked at it for over a year, (thanks for the reminder). Its still running (486/66, 64M ram, and uptime over 7 years :-) Gary ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PrePaid Application!!!!!
Bartosz Jozwiak wrote: Hello, Here in our office we are testing Asterisk. My collage Igor created to Asterisk PrePaid application with Postgresql. It is not in Perl. We would like to release it to the group as soon as it will work ok. It will have authentication, different rates for users, different rates for destinations and so on. Is there anybody who would like to improve it ? -- Bart Of course. This is the meaning of releasing :) For example i have strong knowledge + expirience writing application that using PgSQL ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Parking and Paid Digium software modifications
Is there an underlying reason you want to do this? Because if a call is already parked on 701 and you transfer another call to 701 to park it, both callers would be connected. Actually I have to agree with Matt; I would like to be able to specify where it's parked and get a busy if I try to park a call where there is already one waiting. That's how the old KSU worked anyway. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
I'm curently looking into using SER to front end SIP calls for Asterisk. Basicaly all SIP users would register with SER not Asterisk and then Asterisk and SER exchange registrations. SER is a very capable SIP router, much more sophisticated than Asterisk as it can look inside packets and route based on what it finds or even re-write packets based on user specified logic. SER is GPL'd and has very good user documentation. Don't know how well the above will work. The claim by the authors or SER that it can handle thousands of calls per second is quite impressive One other nice feature is that SER users can set up their own SIP accounts using a web interface and not needing to edit *.conf files. See here for details http://www.iptel.org/ser/ = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Parking and Paid Digium software modifications
Most PBX do park the way your old KSU system did. As a matter of fact Asterisk is the only PBX I have ever seen that parks the way it does. If given a choice my uses would use the normal way. And I would be happy not to here the question can you speed up her talking? LOL Andrew Kohlsmith wrote: Is there an underlying reason you want to do this? Because if a call is already parked on 701 and you transfer another call to 701 to park it, both callers would be connected. Actually I have to agree with Matt; I would like to be able to specify where it's parked and get a busy if I try to park a call where there is already one waiting. That's how the old KSU worked anyway. Regards, Andrew ___ 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
RE: [Asterisk-Users] Call Parking and Paid Digium software modifi cations
That is how many old PBX phone systems work and it is that way our users are used to working with the phone system. Another issue with the way Asterisk callparking currently works is that there is only one call-park orbit, you cannot use a different set of numbers for a different call park instance(i.e. 700 goes to 701-720 AND 740 goes to 741-750). We also have several Grandstream phones which cannot use the asterisk implementation of call-parking because they cannot hear the extension that Asterisk chose for their call to go to. The best solution for my company is to be like the old systems and allow people to define the exact extension that they park their calls on, and if that park extension is busy it would give a busy signal. MATT--- -Original Message- From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] Sent: Monday, October 13, 2003 6:59 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Call Parking and Paid Digium software modifications Is there an underlying reason you want to do this? Because if a call is already parked on 701 and you transfer another call to 701 to park it, both callers would be connected. Actually I have to agree with Matt; I would like to be able to specify where it's parked and get a busy if I try to park a call where there is already one waiting. That's how the old KSU worked anyway. Regards, Andrew ___ 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
Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
I'm curently looking into using SER to front end SIP calls for Asterisk. Basicaly all SIP users would register with SER not Asterisk and then Asterisk and SER exchange registrations. SER is a very capable SIP router, much more sophisticated than Asterisk as it can look inside packets and route based on what it finds or even re-write packets based on user specified logic. SER is GPL'd and has very good user documentation. Don't know how well the above will work. The claim by the authors or SER that it can handle thousands of calls per second is quite impressive One other nice feature is that SER users can set up their own SIP accounts using a web interface and not needing to edit *.conf files. See here for details http://www.iptel.org/ser/ = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK SER is an excellent option as a front end to Asterisk. It is a true SIP proxy, whereas Asterisk is a hybrid, and SIP has not been the primary focus of Asterisk development. In fact, Asterisk's SIP implementation is very limited (though it is extremely pragmatic.) However, moving to SER does not solve any of the issues about the proxy being behind a NAT, and I believe that SER will have the same problems (though I could be wrong on this; I haven't experimented with SER's ability to work from behind a NAT.) SIP clients work well enough behind NAT (most of them, anyway) but the servers are a different story. I really like SER's third-party addons for account administration; Asterisk is significantly more complex, and probably would not be as easily converted to such a front end. In fact, SER has a very complex routing/scripting language that is not easily administered with a web front end, so I think that SER and Asterisk suffer from the same problems. If someone were to come up with a simple way to administer voicemail.conf and sip.conf from a web tool, that would go far to making Asterisk a bit more user-accessible... JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PRI/E1: machine freeze/dies after a few calls
Hi Thomas- I didn't look closely at your shell script, but I wrote something similar in Perl (and used shell to start each instance of it). I had a few problems too with a similar setup (although no machine lockups) * You are running quite a slow machine to run this script on many lines at once - I found that I needed a P4, 2.4GHz to keep up with 120 channels simultaneously (I had one system to send and one to receive, and very short calls - 3 seconds). How many instances are you running? Are you doing mySQL call logging? * I found that I could only initiate about 18 calls at exactly the same moment without getting failed outbound call errors from asterisk, so I ended up staggering the start times a little. * With lots of new calls, I had tons of framing errors on the receiving end (and occasional D channel restarts) when routing calls through my DMS100 switch - do you have problems like this? I think this problem is specific to the Nortel switch however. Suggest starting with -c and routing all output to a log file...?? regards Scott Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thomas Haeger Sent: Monday, October 13, 2003 8:00 PM To: Asterisk User Subject: [Asterisk-Users] PRI/E1: machine freeze/dies after a few calls Hi all, inside my * is a E400P. The machine is a PII 400Mhz with 256MB Ram. OS is Debian woody. * is the newest cvs co. I have written a little callgen script which make outgoing calls through my *: #! /bin/sh set -e n=$1# Nummer anz=$2 # Anzhal der Versuche anz2=$3 # Kanäle sle=$4 # Timeout bis zum nächsten Versuch if [ -z $4 ]; then sle=0 fi s=1 i=1 while [ $s -le $anz ];do echo $s try started... while [ $i -le $anz2 ];do echo -e Channel: Zap/g1/$n$i\nMaxRetries: 0\nContext: callgen\nExtension: 1\nPriority: 1\nCallerid:334778\n /var/spool/asterisk/outgoing/call.$i.$s sleep 2 i=$((i + 1)) done i=1 echo sleep for $sle sec. sleep $sle s=$((s + 1)) done The calls goes out over the first two ports and through a pri switch (teles) they come back at the other two ports (3 and 4). But after a few calls my machine is completly freezed! So that i have to restart my machine. Here're my extension.conf, zapata.conf and zaptel.conf: extension.conf: [pri1] exten = _X.,1,SetAccount(pritest) exten = _X.,2,Answer exten = _X.,3,Wait(15) exten = _X.,4,Hangup [pri2] exten = _X.,1,SetAccount(pritest) exten = _X.,2,Answer exten = _X.,3,Wait(15) exten = _X.,4,Hangup [pri3] exten = _X.,1,SetAccount(pritest) exten = _X.,2,Answer exten = _X.,3,Wait(60) exten = _X.,4,Hangup [pri4] exten = _X.,1,SetAccount(pritest) exten = _X.,2,Answer exten = _X.,3,Wait(60) exten = _X.,4,Hangup [callgen] exten = 1,1,Wait(90) zapata.conf: ; ; Zapata telephony interface ; ; Configuration file [channels] pridialplan=local switchtype=euroisdn busydetect=yes callprogress=no echocancel=yes echocancelwhenbridged=yes ;callwaitingcallerid=no ;callwaiting=no signalling=pri_net group=1 context=pri1 channel = 1-15,17-31 channel =32-46,48-62 signalling=pri_net group=3 context=pri3 channel = 63-77,79-93 channel = 94-108,110-124 zaptel.conf span=1,0,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 span=3,0,0,ccs,hdb3,crc4 span=4,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 bchan=63-77,79-93 dchan=78 bchan=94-108,110-124 dchan=109 loadzone = fr defaultzone=fr Thanks for your help. Regards, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ 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
Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
On 13-10 17:11, John Todd wrote: [...] SER is an excellent option as a front end to Asterisk. It is a true SIP proxy, whereas Asterisk is a hybrid, and SIP has not been the primary focus of Asterisk development. In fact, Asterisk's SIP implementation is very limited (though it is extremely pragmatic.) However, moving to SER does not solve any of the issues about the proxy being behind a NAT, and I believe that SER will have the same problems (though I could be wrong on this; I haven't experimented with SER's ability to work from behind a NAT.) SIP clients work well enough behind NAT (most of them, anyway) but the servers are a different story. SER can can become very helpful when it is run in the public internet and clients are behind NATs. For this case SER contains many NAT helping functions that can rewrite header fields, test if a client comes from behind a NAT, ping clients behind NATs (to keep the NAT binding open) and force RTP proxy usage when necesary. Along with RTP proxy SER can help any *symmetric* SIP user agent to get through NAT. (A symmetric SIP user agent is a user agent that uses the same source port for receiving signalling and media as for sending them. Vast majority of SIP user agents as of today is symmetric, including Windows Messenger, Cisco phones, Grandstream phone a.s.o.). There is also support for proxy behind NAT, but it is mostly untested yet. Jan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoiceMail fromstring?
I just tested fromstring and emailbody with voicemail2 and a farily new code and it's working. I don't know what you're doing wrong ... but something for sure. regards Martin On Mon, 13 Oct 2003, John Todd wrote: I would recommend then doing grep fromstring /usr/src/asterisk/apps/app_voicemail2.c Martin On Fri, 19 Sep 2003, Ben Bloomberg wrote: I'm having tons of trouble getting the fromstring to work in voicemail.conf. I've tried both voicemail and voicemail2 but the emails still seem to be coming from asterisk pbx. Has anyone had any luck with this? [snip] Martin - I examined the source, but I am still un-enlightened. :-) I cannot get fromstring or emailbody working reliably. Even with the minimalist settings below, the header or body did not change (other than serveremail which seems to be set appropriately.) Interestingly and perhaps as an additional problem, the timezones also don't seem to work correctly in the voicemail message, either - the time in the email message is Eastern time (the TZ to which that server is set.) My CVS is Asterisk CVS-10/13/03-18:38:10. What I am doing incorrectly? JT [general] format=wav [EMAIL PROTECTED] attach=yes fromstring=Foo emailbody=New vm now [zonemessages] eastern=US/NewYork|'vm-received' Q 'digits/at' IMp central=US/Central|'vm-received' Q 'digits/at' IMp mountain=US/Mountain|'vm-received' Q 'digits/at' IMp pacific=US/Pacific|'vm-received' Q 'digits/at' IMp [default] 2413669780 = ,john todd,[EMAIL PROTECTED],,|tz=pacific A message left in that mailbox results in: Date: Mon, 13 Oct 2003 18:53:49 -0400 From: Asterisk PBX [EMAIL PROTECTED] To: john todd [EMAIL PROTECTED] Subject: [PBX]: New message 2 in mailbox 2413669780 Dear john todd: Just wanted to let you know you were just left a 0:01 long message (number 2) in mailbox 2413669780 from 2155821314, on Monday, October 13, 2003 at 06:53:49 PM so you might want to check it when you get a chance. Thanks! --Asterisk Content-Type: audio/x-wav; name=msg0002.wav Content-Description: Voicemail sound attachment. Content-Disposition: attachment; filename=msg0002.wav ___ 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
RE: [Asterisk-Users] X100P Echo Problems..What's going to happen?
John: I have been around voice over data packets for quite a few years and I am still to see the perfect system that works identical to circuit switching 100% of the time. My opinion is that there is a lot more to the story than just parameters. Packet loses, double compressions, faulty routers, bandwidth, analog to digital and so on can get in the way. On the other hand, if your customer understand the benefits, and I mean more than cost, and can leave with 80% perfect, then you will be able to understand why a lot of companieshaveopted for VoIP (or ATM or Frame Relay). Regards, Uriel-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of John MSent: Monday, October 13, 2003 1:41 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] X100P Echo Problems..What's going to happen?Importance: High Ive read and experienced the echo problems with the X100P. Is Digium going to fix the problem or refund our money? I want to see this work because myself and other small companies out there use analog lines. I would trade up to T1 but that requires me to have at least 9 lines. If I did trade up, do the T1 cards work perfectly with no echo at all? I get echo with my directly connected computer using Xten SIP. No matter with all the suggestions to change the parameters, it still has echo. Does anyone have the T1 and have no problems at all? I would surely appreciate you experiences. Whats my option to get this too work flawlessly? John
RE: [Asterisk-Users] No sound with SIP Phones on the Internet
Dunca: I am not sure I understand your statemnet. SIP devices (UA) on the other side of the Internet behnid a NAT communicate to * on the public Internet. Then this Asterisk connects to other Asterisks (via IAX) that can be behind Firwalls (or NATS). am I understanding correctly? Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of duncan Sent: Monday, October 13, 2003 12:25 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] No sound with SIP Phones on the Internet This is bull... I can't believe that... Must be a solution... sip is very tricky to get working behind firewalls. sip clients work quite well with nat, just make sure nat=yes is in the sip profile in sip.conf my solution has always been to put an asterisk box behind the firewall and make all the sip clients connect to that, then IAX out of the firewall to the other machines. i spent a few days trying unsuccessfully to find a decent sip proxy that worked the way i wanted and decided that the asterisk solution was much better. duncan ___ 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
RE: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
Chris: I am glad to see someone else asking the same question I have been asking myself. As soon as I get my public IP address, I will install SER on the public side and Asterisk behind a NAT (with dynamic IP) to see if I can get around problems I have when my SIP (UA) behind their own NAT on the other side of my Internet connection. If you make any progress, please share. I will do the same. Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Albertson Sent: Monday, October 13, 2003 7:49 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet) I'm curently looking into using SER to front end SIP calls for Asterisk. Basicaly all SIP users would register with SER not Asterisk and then Asterisk and SER exchange registrations. SER is a very capable SIP router, much more sophisticated than Asterisk as it can look inside packets and route based on what it finds or even re-write packets based on user specified logic. SER is GPL'd and has very good user documentation. Don't know how well the above will work. The claim by the authors or SER that it can handle thousands of calls per second is quite impressive One other nice feature is that SER users can set up their own SIP accounts using a web interface and not needing to edit *.conf files. See here for details http://www.iptel.org/ser/ = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.yahoo.com ___ 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
RE: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
John: are you aware of any documentation on how to configre SER to be a front-end to Asterisk? I suspect it is very inexpensive to put a SER server in a hosting facility to forward traffic to multiple Asterisks based on Least Cost Routing. My problem is that my experience is with Asterisk and not with SER. Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Todd Sent: Monday, October 13, 2003 8:11 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet) I'm curently looking into using SER to front end SIP calls for Asterisk. Basicaly all SIP users would register with SER not Asterisk and then Asterisk and SER exchange registrations. SER is a very capable SIP router, much more sophisticated than Asterisk as it can look inside packets and route based on what it finds or even re-write packets based on user specified logic. SER is GPL'd and has very good user documentation. Don't know how well the above will work. The claim by the authors or SER that it can handle thousands of calls per second is quite impressive One other nice feature is that SER users can set up their own SIP accounts using a web interface and not needing to edit *.conf files. See here for details http://www.iptel.org/ser/ = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK SER is an excellent option as a front end to Asterisk. It is a true SIP proxy, whereas Asterisk is a hybrid, and SIP has not been the primary focus of Asterisk development. In fact, Asterisk's SIP implementation is very limited (though it is extremely pragmatic.) However, moving to SER does not solve any of the issues about the proxy being behind a NAT, and I believe that SER will have the same problems (though I could be wrong on this; I haven't experimented with SER's ability to work from behind a NAT.) SIP clients work well enough behind NAT (most of them, anyway) but the servers are a different story. I really like SER's third-party addons for account administration; Asterisk is significantly more complex, and probably would not be as easily converted to such a front end. In fact, SER has a very complex routing/scripting language that is not easily administered with a web front end, so I think that SER and Asterisk suffer from the same problems. If someone were to come up with a simple way to administer voicemail.conf and sip.conf from a web tool, that would go far to making Asterisk a bit more user-accessible... JT ___ 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
RE: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Todd Sent: Monday, October 13, 2003 8:11 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet) I'm curently looking into using SER to front end SIP calls for Asterisk. Basicaly all SIP users would register with SER not Asterisk and then Asterisk and SER exchange registrations. SER is a very capable SIP router, much more sophisticated than Asterisk as it can look inside packets and route based on what it finds or even re-write packets based on user specified logic. SER is GPL'd and has very good user documentation. Don't know how well the above will work. The claim by the authors or SER that it can handle thousands of calls per second is quite impressive One other nice feature is that SER users can set up their own SIP accounts using a web interface and not needing to edit *.conf files. See here for details http://www.iptel.org/ser/ = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK SER is an excellent option as a front end to Asterisk. It is a true SIP proxy, whereas Asterisk is a hybrid, and SIP has not been the primary focus of Asterisk development. In fact, Asterisk's SIP implementation is very limited (though it is extremely pragmatic.) However, moving to SER does not solve any of the issues about the proxy being behind a NAT, and I believe that SER will have the same problems (though I could be wrong on this; I haven't experimented with SER's ability to work from behind a NAT.) SIP clients work well enough behind NAT (most of them, anyway) but the servers are a different story. I really like SER's third-party addons for account administration; Asterisk is significantly more complex, and probably would not be as easily converted to such a front end. In fact, SER has a very complex routing/scripting language that is not easily administered with a web front end, so I think that SER and Asterisk suffer from the same problems. If someone were to come up with a simple way to administer voicemail.conf and sip.conf from a web tool, that would go far to making Asterisk a bit more user-accessible... JT At 11:26 PM -0400 10/13/03, Uriel Carrasquilla wrote: From: Uriel Carrasquilla [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet) Reply-To: [EMAIL PROTECTED] Date: Mon, 13 Oct 2003 23:26:59 -0400 John: are you aware of any documentation on how to configre SER to be a front-end to Asterisk? I suspect it is very inexpensive to put a SER server in a hosting facility to forward traffic to multiple Asterisks based on Least Cost Routing. My problem is that my experience is with Asterisk and not with SER. Uriel Uriel - 1) Please stop top-posting. 2) I'm afraid I don't have any data on specifics of creating a front-end. I know how to do it, but my time these days is spent writing lots of other projects that I have been doing. :-) I would suggest you get SER and set it up - it's quite easy, and the documentation on SER itself is very well written, and if you have a good idea of how SIP works you should be able to patch together an appropriate system. However, if you aren't 100% familiar with how SIP works, I would stick to just an Asterisk system; SER doesn't allow for any of the shortcuts that Asterisk has. 3) Use Google and do some searching. I found some quick links with a few of the keywords that would seem obvious, but I don't have enough time to review them... JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI solution to Grandstream BT102 call waiting problem
I'm trying to fix a problem with the GrandStream Budgetone 102. I've been reading the
source code, mailing lists and other resources. Here's the scenario and the approach
I have been pursuing. I'm having some problems with the AGI calls and I hope someone
can give me some clarification.
PSTN --- T1,PRI * --- Grandstream BT 102 (12)
--- TDM400 (1) Fax machine
End user requires that an inbound call ring 4 of the BT 102 phones so that any
available employee may answer. Generally, this works very well and the only problem I
have is when another call comes in. In this case the Grandstream BT 102 rings very
loudly in the ear piece and obliterates the conversation. The external party notices
that the audio is cut off while the BT 102 is ringing.
The solution I am trying to develop to work around the problem is to use an AGI script
to check each of the extensions in the group of phones that are supposed to be rung.
I want to remove extensions that are presently in a call and not ring them with the
new call. Also, I want to perform this test every 10 seconds so that I may include
any extension back into the group of phones to ring if their previous call has ended.
I plan to implement a variable and increment it in a loop using gotoif to provide
three or four 10 second trials for the dial. If no one answers, I'll send it to voice
mail.
I have tried using ChanIsAvail and Channel status to see if I can detect when the Sip
phone is busy. In the case of the ChanIsAvail, it doesn't matter if the phone is busy
or not, it will still return the channel as available. Maybe the definition of
channel is available does not have anything to do if it is in a call or not. In the
case of the Channel Status, it always returns 201 on the Sip channel. Actually, I'm
using the Asterisk Perl Modules by James Golovich so the
$AGI-channel_status('Zap/1-1') returns 4 and the $AGI-channel_status('Sip/2400')
returns -1.
I think that a major problem with the Channel Status is that the Sip channel is not
being correctly provided. Since it seems to work with the Zap channel. The code is
walking the channels to do a strcmp for an exact match. I'm lost to find out what the
Sip channel designator should look like.
It appears that the ChanIsAvail would be the correct call to make for this purpose.
Last, I can't get the options to work with the $AGI-exed('Dial', $newvar , '30,t');
command. It seems to ignore the options, so, I can't tell the dial command how long
to ring and to allow the called extension to transfer.
TIA, Walker
Here are some snips from my conf files and agi script:
myagi.agi
...
for $i ( split /,/,$ARGV[0] ) {
if ( $AGI-exec('ChanIsAvail', $i) == 0 ) {
if ( $count++ 0 ) { $newvar .= \ };
$newvar .= $i;
$result = $AGI-channel_status($i); # always returns -1
print STDERR $result\n;
}
}
$result = $AGI-channel_status('Zap/1-1'); # test this with Zap
print STDERR $result\n; # always returns 4
$AGI-exec('Dial', $newvar , '30,t'); # this is supposed to dial the
# extensions that are not busy
...
sip.conf
...
[2400] ; Grandstream Phone
context=intern
type=friend
insecure=yes
host=dynamic
permit=192.168.254.0/255.255.255.0
mailbox=2400
dtmfmode=inband
canreinvite=no
nat=no
...
extensions.conf
...
PHONE2=SIP/2400
PHONE3=SIP/2410
RECEPTION=${PHONE2},${PHONE3}
...
exten = 2200,1,AGI(myagi.agi,${RECEPTION})
...
console (asterisk -vvvc)
...
-- Executing AGI(Zap/1-1, myagi.agi|SIP/2400,SIP/2410) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/myagi.agi
arg1 = SIP/2400,SIP/2410
Channel Status:
-- AGI Script Executing Application: (ChanIsAvail) Options: (SIP/2400)
-1
-- AGI Script Executing Application: (ChanIsAvail) Options: (SIP/2410)
-1
4
-- AGI Script Executing Application: (Dial) Options: (SIP/2400SIP/2410)
-- Called 2400
-- Called 2410
-- SIP/2400-3320 is ringing
-- SIP/2410-a8ca is ringing
...
--
DataCrest, Inc. -- Technically Superior **
Walker Haddock http://www.datacrest.com
DataCrest, Inc.e-mail: [EMAIL PROTECTED]
1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589
Birmingham, AL 35216 fax: 1-205-823-7838
***
--
DataCrest, Inc. -- Technically Superior **
Walker Haddock http://www.datacrest.com
DataCrest, Inc.e-mail: [EMAIL PROTECTED]
1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589
Birmingham, AL 35216 fax: 1-205-823-7838
***
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[Asterisk-Users] AGI solution to Grandstream BT102 call waiting problem
I'm trying to fix a problem with the GrandStream Budgetone 102. I've been reading the
source code, mailing lists and other resources. Here's the scenario and the approach
I have been pursuing. I'm having some problems with the AGI calls and I hope someone
can give me some clarification.
PSTN --- T1,PRI * --- Grandstream BT 102 (12)
--- TDM400 (1) Fax machine
End user requires that an inbound call ring 4 of the BT 102 phones so that any
available employee may answer. Generally, this works very well and the only problem I
have is when another call comes in. In this case the Grandstream BT 102 rings very
loudly in the ear piece and obliterates the conversation. The external party notices
that the audio is cut off while the BT 102 is ringing.
The solution I am trying to develop to work around the problem is to use an AGI script
to check each of the extensions in the group of phones that are supposed to be rung.
I want to remove extensions that are presently in a call and not ring them with the
new call. Also, I want to perform this test every 10 seconds so that I may include
any extension back into the group of phones to ring if their previous call has ended.
I plan to implement a variable and increment it in a loop using gotoif to provide
three or four 10 second trials for the dial. If no one answers, I'll send it to voice
mail.
I have tried using ChanIsAvail and Channel status to see if I can detect when the Sip
phone is busy. In the case of the ChanIsAvail, it doesn't matter if the phone is busy
or not, it will still return the channel as available. Maybe the definition of
channel is available does not have anything to do if it is in a call or not. In the
case of the Channel Status, it always returns 201 on the Sip channel. Actually, I'm
using the Asterisk Perl Modules by James Golovich so the
$AGI-channel_status('Zap/1-1') returns 4 and the $AGI-channel_status('Sip/2400')
returns -1.
I think that a major problem with the Channel Status is that the Sip channel is not
being correctly provided. Since it seems to work with the Zap channel. The code is
walking the channels to do a strcmp for an exact match. I'm lost to find out what the
Sip channel designator should look like.
It appears that the ChanIsAvail would be the correct call to make for this purpose.
Last, I can't get the options to work with the $AGI-exed('Dial', $newvar , '30,t');
command. It seems to ignore the options, so, I can't tell the dial command how long
to ring and to allow the called extension to transfer.
TIA, Walker
Here are some snips from my conf files and agi script:
myagi.agi
...
for $i ( split /,/,$ARGV[0] ) {
if ( $AGI-exec('ChanIsAvail', $i) == 0 ) {
if ( $count++ 0 ) { $newvar .= \ };
$newvar .= $i;
$result = $AGI-channel_status($i); # always returns -1
print STDERR $result\n;
}
}
$result = $AGI-channel_status('Zap/1-1'); # test this with Zap
print STDERR $result\n; # always returns 4
$AGI-exec('Dial', $newvar , '30,t'); # this is supposed to dial the
# extensions that are not busy
...
sip.conf
...
[2400] ; Grandstream Phone
context=intern
type=friend
insecure=yes
host=dynamic
permit=192.168.254.0/255.255.255.0
mailbox=2400
dtmfmode=inband
canreinvite=no
nat=no
...
extensions.conf
...
PHONE2=SIP/2400
PHONE3=SIP/2410
RECEPTION=${PHONE2},${PHONE3}
...
exten = 2200,1,AGI(myagi.agi,${RECEPTION})
...
console (asterisk -vvvc)
...
-- Executing AGI(Zap/1-1, myagi.agi|SIP/2400,SIP/2410) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/myagi.agi
arg1 = SIP/2400,SIP/2410
Channel Status:
-- AGI Script Executing Application: (ChanIsAvail) Options: (SIP/2400)
-1
-- AGI Script Executing Application: (ChanIsAvail) Options: (SIP/2410)
-1
4
-- AGI Script Executing Application: (Dial) Options: (SIP/2400SIP/2410)
-- Called 2400
-- Called 2410
-- SIP/2400-3320 is ringing
-- SIP/2410-a8ca is ringing
...
--
DataCrest, Inc. -- Technically Superior **
Walker Haddock http://www.datacrest.com
DataCrest, Inc.e-mail: [EMAIL PROTECTED]
1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589
Birmingham, AL 35216 fax: 1-205-823-7838
***
--
DataCrest, Inc. -- Technically Superior **
Walker Haddock http://www.datacrest.com
DataCrest, Inc.e-mail: [EMAIL PROTECTED]
1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589
Birmingham, AL 35216 fax: 1-205-823-7838
***
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[Asterisk-Users] H323 ID's
Hello, Is there any way to pass an H323 ID (resembles a sip [EMAIL PROTECTED]) to an h323 gateway? Thank you in advance for your suggestions! Regards, Christopher J. Wolff, VP, CIO Broadband Laboratories http://www.bblabs.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoiceMail fromstring?
However, the timezone is still not straight in the message body.
${VM_DATE} doesn't seem to use the timezone matching routines defined
by the user's tz= setting.
Well it's the task for those who add features to have a global-system
thinking. The emailbody was added way before the timezones ...
Also, there seems to be a character limit for the length of
emailbody= that is a bit short - I get the last part of my messages
chopped off at a predictable point (seems to be around the 500th
character of the emailbody= line that it gets snipped.)
That can be easily changes since the static array is used.
Martin
JT
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[Asterisk-Users] MGCP Gateway (Dlink DG104s)
Has anyone gotten 3 way calling to work? There seems to be no way to swap to the other call and sometimes the unit will generate the call waiting tone ever second. It also seems that if you try to flash the call and then hang up you have to pick up the phone, flash back to the first call and than hang up in order to hangup the first call.
Re: [Asterisk-Users] VoiceMail fromstring?
On Monday 13 October 2003 22:18, Martin Pycko wrote:
However, the timezone is still not straight in the message body.
${VM_DATE} doesn't seem to use the timezone matching routines
defined by the user's tz= setting.
Well it's the task for those who add features to have a global-system
thinking. The emailbody was added way before the timezones ...
You could have just said it was a bug, instead of insulting a
contributing developer. Patch attached.
Also, there seems to be a character limit for the length of
emailbody= that is a bit short - I get the last part of my
messages chopped off at a predictable point (seems to be around the
500th character of the emailbody= line that it gets snipped.)
That can be easily changes since the static array is used.
Actually, no, a static array is not used. The code assumes that a
minimum of twice the length of the email body (or 100 minimum) is of
sufficient length. But it's easily changed.
int vmlen = strlen(emailbody)*2;
if (vmlen 20)
vmlen = 100;
passdata = alloca(vmlen);
bzero( passdata, vmlen );
-Tilghman
Index: apps/app_voicemail2.c
===
RCS file: /usr/cvsroot/asterisk/apps/app_voicemail2.c,v
retrieving revision 1.56
diff -u -r1.56 app_voicemail2.c
--- apps/app_voicemail2.c 4 Oct 2003 22:08:02 - 1.56
+++ apps/app_voicemail2.c 14 Oct 2003 04:42:47 -
@@ -25,6 +25,7 @@
#include asterisk/app.h
#include asterisk/manager.h
#include asterisk/dsp.h
+#include asterisk/localtime.h
#include stdlib.h
#include errno.h
#include unistd.h
@@ -631,7 +632,7 @@
return 1;
}
-static int sendmail(char *srcemail, char *email, char *name, int msgnum, char *mailbox, char *callerid, char *attach, char *format, long duration, int attach_user_voicemail)
+static int sendmail(char *srcemail, struct ast_vm_user *vmu, int msgnum, char *mailbox, char *callerid, char *attach, char *format, long duration, int attach_user_voicemail)
{
FILE *p;
char date[256];
@@ -642,6 +643,8 @@
char dur[256];
time_t t;
struct tm tm;
+ struct vm_zone *the_zone = NULL;
+
if (!strcmp(format, wav49))
format = WAV;
ast_log(LOG_DEBUG, Attaching file '%s', format '%s', uservm is '%d', global is %d\n, attach, format, attach_user_voicemail, attach_voicemail);
@@ -655,7 +658,25 @@
}
snprintf(dur, sizeof(dur), %ld:%02ld, duration / 60, duration % 60);
time(t);
- localtime_r(t,tm);
+
+ /* Does this user have a timezone specified? */
+ if (strlen(vmu-zonetag)) {
+ /* Find the zone in the list */
+ struct vm_zone *z;
+ z = zones;
+ while (z) {
+if (!strcmp(z-name, vmu-zonetag)) {
+ the_zone = z;
+ break;
+}
+z = z-next;
+ }
+ }
+
+ if (the_zone)
+ ast_localtime(t,tm,the_zone-timezone);
+ else
+ ast_localtime(t,tm,NULL);
strftime(date, sizeof(date), %a, %d %b %Y %H:%M:%S %z, tm);
fprintf(p, Date: %s\n, date);
@@ -663,7 +684,7 @@
fprintf(p, From: %s %s\n, fromstring, who);
else
fprintf(p, From: Asterisk PBX %s\n, who);
- fprintf(p, To: %s %s\n, name, email);
+ fprintf(p, To: %s %s\n, vmu-fullname, vmu-email);
if( *emailtitle)
{
@@ -696,7 +717,7 @@
vmlen = 100;
passdata = alloca(vmlen);
bzero( passdata, vmlen );
-pbx_builtin_setvar_helper(ast, VM_NAME, name);
+pbx_builtin_setvar_helper(ast, VM_NAME, vmu-fullname);
pbx_builtin_setvar_helper(ast, VM_DUR, dur);
sprintf(passdata,%d,msgnum);
pbx_builtin_setvar_helper(ast, VM_MSGNUM, passdata);
@@ -711,7 +732,7 @@
fprintf(p, Dear %s:\n\n\tJust wanted to let you know you were just left a %s long message (number %d)\n
in mailbox %s from %s, on %s so you might\n
- want to check it when you get a chance. Thanks!\n\n\t\t\t\t--Asterisk\n\n, name,
+ want to check it when you get a chance. Thanks!\n\n\t\t\t\t--Asterisk\n\n, vmu-fullname,
dur, msgnum + 1, mailbox, (callerid ? callerid : an unknown caller), date);
}
if (attach_user_voicemail) {
@@ -733,7 +754,7 @@
return 0;
}
-static int sendpage(char *srcemail, char *pager, int msgnum, char *mailbox, char *callerid, long duration)
+static int sendpage(char *srcemail, char *pager, int msgnum, char *mailbox, char *callerid, long duration, struct ast_vm_user *vmu)
{
FILE *p;
char date[256];
@@ -742,6 +763,7 @@
char dur[256];
time_t t;
struct tm tm;
+ struct vm_zone *the_zone = NULL;
p = popen(SENDMAIL, w);
if (p) {
@@ -753,7 +775,26 @@
}
snprintf(dur, sizeof(dur), %ld:%02ld, duration / 60, duration % 60);
time(t);
- localtime_r(t,tm);
+
+ /* Does this user have a timezone specified? */
+ if (strlen(vmu-zonetag)) {
+ /* Find the zone in the list */
+ struct vm_zone *z;
+ z = zones;
+ while (z) {
+if (!strcmp(z-name, vmu-zonetag)) {
+ the_zone = z;
+ break;
+}
+z = z-next;
+ }
+ }
+
+ if (the_zone)
+
Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
On Monday 13 October 2003 22:26, Uriel Carrasquilla wrote:
John:
are you aware of any documentation on how to configre SER to be a front-end
to Asterisk?
Hi Uriel,
At TeleSIP we run a cluster of several geographically distributed SER Servers
that hande all our SIP Routing. SER is a robust, fast and stable platform
which has worked flawlessly for us. We use * as our company PBX and PSTN
Gateway. Basically what you need to do is to device a numbering plan so that
SERs routing logic can forward the call to * when it needs to.
For example in ser.cfg you could put something like this:
#
###PSTN ACCESS###
#
if (method==INVITE) {
if (uri=~sip:[EMAIL PROTECTED]) {
log(1, This is a Long Distance Call\n);
route(6);
break;
};
};
.
.
.
route[6] {
rewritehostport(your_asterisk_box:5050);
if (!t_relay()) {
sl_reply_error();
};
}
Andres
http://www.telesip.net
I suspect it is very inexpensive to put a SER server in a hosting facility
to forward traffic to multiple Asterisks based on Least Cost Routing.
My problem is that my experience is with Asterisk and not with SER.
Uriel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Todd
Sent: Monday, October 13, 2003 8:11 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on
the Internet)
I'm curently looking into using SER to front end SIP calls for
Asterisk.
Basicaly all SIP users would register with SER not Asterisk and then
Asterisk and SER exchange registrations.
SER is a very capable SIP router, much more sophisticated than Asterisk
as it can look inside packets and route based on what it finds or even
re-write packets based on user specified logic.
SER is GPL'd and has very good user documentation. Don't know how well
the above will work. The claim by the authors or SER that it can
handle thousands of calls per second is quite impressive
One other nice feature is that SER users can set up their own SIP
accounts using a web interface and not needing to edit *.conf files.
See here for details http://www.iptel.org/ser/
=
Chris Albertson
Home: 310-376-1029 [EMAIL PROTECTED]
Cell: 310-990-7550
Office: 310-336-5189 [EMAIL PROTECTED]
KG6OMK
SER is an excellent option as a front end to Asterisk. It is a
true SIP proxy, whereas Asterisk is a hybrid, and SIP has not been
the primary focus of Asterisk development. In fact, Asterisk's SIP
implementation is very limited (though it is extremely pragmatic.)
However, moving to SER does not solve any of the issues about the
proxy being behind a NAT, and I believe that SER will have the same
problems (though I could be wrong on this; I haven't experimented
with SER's ability to work from behind a NAT.) SIP clients work
well enough behind NAT (most of them, anyway) but the servers are a
different story.
I really like SER's third-party addons for account administration;
Asterisk is significantly more complex, and probably would not be as
easily converted to such a front end. In fact, SER has a very
complex routing/scripting language that is not easily administered
with a web front end, so I think that SER and Asterisk suffer from
the same problems. If someone were to come up with a simple way to
administer voicemail.conf and sip.conf from a web tool, that would go
far to making Asterisk a bit more user-accessible...
JT
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[Asterisk-Users] Problem with SIP authentication
Hi List, After going through mailing list and manual of asterisk, I still could not properly get authenticated with my SIP UA to asterisk. i m using a username for UA "12321" and following are SIP.conf file user params [12321]type=friendusername=12321host=dynamicsecret=ccartacontext=defaultmailbox=1234,2345 ; Mailbox for message waiting indicator [7]type=friendusername=7host=dynamicsecret=atracccontext=defaultmailbox=1234,2345 m trying with SIPPS UA, that gets status "Acquired", not "Registered", Can anyone give any idea about it? I tried same with X-Lite, didnt work. Sip debug messages are pasted below. Best Regards, JF Sip read:REGISTER sip:192.168.100.71 SIP/2.0Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992800961-40de3e5f192.168.100.66From: sip:[EMAIL PROTECTED];tag=3b2cf0baTo: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]Contact: ccarta sip:[EMAIL PROTECTED]:5062;expires=600;q=0.500Expires: 600CSeq: 1 REGISTERContent-Length: 0User-Agent: Ahead SIPPS IP Phone Version 2.0.42.13 10 headers, 0 linesUsing latest request as basis requestSending to 192.168.100.66 : 5062 (non-NAT)Transmitting (no NAT):SIP/2.0 100 TryingVia: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992800961-40de3e5f192.168.100.66From: sip:[EMAIL PROTECTED];tag=3b2cf0baTo: sip:[EMAIL PROTECTED];tag=as648287faCall-ID: [EMAIL PROTECTED]CSeq: 1 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFERContact: sip:[EMAIL PROTECTED]Content-Length: 0 to 192.168.100.66:5062Transmitting (no NAT):SIP/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992800961-40de3e5f192.168.100.66From: sip:[EMAIL PROTECTED];tag=3b2cf0baTo: sip:[EMAIL PROTECTED];tag=as648287faCall-ID: [EMAIL PROTECTED]CSeq: 1 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFERContact: sip:[EMAIL PROTECTED]Proxy-Authenticate: Digest realm="asterisk", nonce="7c7fba4d"Content-Length: 0 to 192.168.100.66:5062Sip read:REGISTER sip:192.168.100.71 SIP/2.0Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992804895-411b471d192.168.100.66From: sip:[EMAIL PROTECTED];tag=3b2d0018To: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]Contact: ccarta sip:[EMAIL PROTECTED]:5062;expires=600;q=0.500Expires: 600CSeq: 2 REGISTERContent-Length: 0Proxy-Authorization: Digest username="12321",realm="asterisk",uri="sip:192.168.100.66",nonce="7c7fba4d",nc="0001",response="b8b1d7fc53eff354dfc31dfa3f800749"User-Agent: Ahead SIPPS IP Phone Version 2.0.42.13 11 headers, 0 linesUsing latest request as basis requestSending to 192.168.100.66 : 5062 (non-NAT)Transmitting (no NAT):SIP/2.0 100 TryingVia: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992804895-411b471d192.168.100.66From: sip:[EMAIL PROTECTED];tag=3b2d0018To: sip:[EMAIL PROTECTED];tag=as648287faCall-ID: [EMAIL PROTECTED]CSeq: 2 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFERContact: sip:[EMAIL PROTECTED]Content-Length: 0 to 192.168.100.66:5062Transmitting (no NAT):SIP/2.0 200 OKVia: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992804895-411b471d192.168.100.66From: sip:[EMAIL PROTECTED];tag=3b2d0018To: sip:[EMAIL PROTECTED];tag=as648287faCall-ID: [EMAIL PROTECTED]CSeq: 2 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFERExpires: 600Contact: sip:[EMAIL PROTECTED];expires=600Date: Tue, 14 Oct 2003 13:46:14 GMTContent-Length: 0 to 192.168.100.66:506211 headers, 2 linesReliably Transmitting:NOTIFY sip:[EMAIL PROTECTED]:5062 SIP/2.0Via: SIP/2.0/UDP 192.168.100.71:5060;branch=z9hG4bK3ecb7a3bFrom: "asterisk" sip:[EMAIL PROTECTED];tag=as3f6e8c0eTo: sip:[EMAIL PROTECTED]:5062Contact: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 102 NOTIFYUser-Agent: Asterisk PBXEvent: message-summaryContent-Type: application/simple-message-summaryContent-Length: 36 Messages-Waiting: noVoicemail: 0/0(no NAT) to 192.168.100.66:5062Sip read:SIP/2.0 200 OKVia: SIP/2.0/UDP 192.168.100.71;branch=z9hG4bK3ecb7a3bFrom: "asterisk" sip:[EMAIL PROTECTED];tag=as3f6e8c0eTo: sip:[EMAIL PROTECTED]:5062;tag=3b302259Call-ID: [EMAIL PROTECTED]CSeq: 102 NOTIFYUser-Agent: Ahead SIPPS IP Phone Version 2.0.42.13Content-Length: 0 8 headers, 0 lines Do you Yahoo!? The New Yahoo! Shopping - with improved product search
Re: [Asterisk-Users] Generating a call with the Manager interface..
Log of real session:
[EMAIL PROTECTED] root]# telnet localhost 5038
Trying 127.0.0.1...
Connected to localhost.
Escape character is '^]'.
Asterisk Call Manager/1.0
action: login
username: joe
secret: bob
Response: Success
Message: Authentication accepted
action: originate
exten: 200
context: stations
channel: SIP/[EMAIL PROTECTED]
Event: Newchannel
Channel: SIP/Jeremy7960-376c
State: Down
Callerid: unknown
Uniqueid: 1066109610.410
Event: Newchannel
Channel: SIP/Jeremy7960-376c
State: Ringing
Callerid: unknown
Uniqueid: 1066109610.410
Event: Newstate
Channel: SIP/Jeremy7960-376c
State: Up
Callerid: unknown
Uniqueid: 1066109610.410
Response: Success
Message: Originate successfully queued
then extensions.conf might look like:
[stations]
exten = 200,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
exten = 200,2, Hangup
Jeremy McNamara
WipeOut wrote:
Hi,
Currently I use call files to automate the generation of calls from
our address book and the resulting call file looks like this..
Channel: SIP/201
WaitTime: 30
Application: Dial
Data: CAPI/4567:5556789
CallerID: Auto Dial 1000
This method works but it not logging the calls to the CDR and there
are a few other issues.. So I wanted to try and do the same thing
using the manager interface in Asterisk.. The problem is that the docs
are a little shy on details..
Does anyone know how I can turn my call file sample into the manager
interface equivalent??
My guess is something like..
Action: Originate
Channel: SIP/201
Timeout: 30
??? (application line)
???(data line)
CallerID: Auto Dial 1000
Thanks..
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