[Asterisk-Users] Réf.: [Asterisk-Users] Call generator
Hi, sipp (http://sipp.sourceforge.net/) seems to be a good app. Take a look at http://www.voip-info.org/wiki-SIPP on the wiki to have more info about it... Basically, there is scenario which are describe there and I personnally generated about 3,000,000 calls before having to restart asterisk and i placed about 90 concurrent calls. Good luck! [EMAIL PROTECTED] a écrit : - Pour: <[EMAIL PROTECTED]> De: "GIBERT Frédéric" <[EMAIL PROTECTED]> Envoyé par: [EMAIL PROTECTED] Date: 23-06-2004 10:46 Objet: [Asterisk-Users] Call generator Hello, Has someone know a good call generator for asterisk including SIP protocol (freeware if possible)? I need to stress a plateform and I don't find any. Thanks by advance. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] A Callback AGI script
Hi there,
I just give you the script (in Python) I have just written in case of
someone would like to implemant this. I think it is more simple than the
one we can see over the net... It uses DISA (security issues ==> limit
access with contexts and the password !!) and CAPI but it should work with
type of channel.
Basically, you ring your asterisk and the line goes down after 1 ring.
Asterisk calls you back with a tonality and you have to enter a password to
dial any extension. The advantage of using DISA is that you can acces any
services you want (=> voicemail...)
If you would like to comment it, do not hesitate !
[EMAIL PROTECTED] cat /var/lib/asterisk/agi-bin/callback.py
#!/usr/bin/env python
import sys
context="incoming"
extension="300"
canal = "CAPI/02xxx89127:"+sys.argv[1]
cf = open("/var/spool/asterisk/outgoing/"+sys.argv[1]+".call",'w+')
cf.write("Channel:"+canal+"\nMaxRetries:2\ncontext:"+context+"\nextension:"+extension+"\npriority:1\nRetryTime:10\nWaitTime:20\n")
[EMAIL PROTECTED] cat /etc/asterisk/extensions.conf | grep -i 300
exten => 300,1,DISA,9641|incoming
exten => 300,2,Hangup
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[Asterisk-Users] Réf.: RE: [Asterisk-Users] SIPP Load testing
You maybe have to create a SIP user called like it is declared in your UAC/UAS xml file. I think it should be 'sipp' or something like that... [EMAIL PROTECTED] a écrit : - Pour: <[EMAIL PROTECTED]> De: "C. Johnson" <[EMAIL PROTECTED]> Envoyé par: [EMAIL PROTECTED] Date: 31-05-2004 08:03 Objet: RE: [Asterisk-Users] SIPP Load testing No, I have not updated since yesterday.. The last * update I did was in March. -cj -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Duane Sent: Monday, May 31, 2004 1:44 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIPP Load testing C. Johnson wrote: > Apparently I'm missing something... Anyone seen this before using SIPP? You updated your asterisk version since yesterday? if so it's the same bug I'm currently trying to work out more details on... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Sipp
Hi there! Does anyone knows how to test Asterisk load with sipp? I am using uac.xml to call a 'playback extensions' via a SIP channel. When I increase the Call rate (about 20cps), I begin to have INVITE/200/BYE retransmissions meanwhile the RedHat box is not loaded at all (made a TOP). Where is the pb? [EMAIL PROTECTED] sipp]# sipp 10.54.196.32 -s 9001 -sf uac.xml -d 100 -i 10.54.196.38 -r 20 Messages Retrans INVITE --> 7152 100<-- 700 200<-- E-RTD 6944 BYE--> 6951 200<-- 690 Thanx! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP seeding
Does anyone knows wether it's possible or not to do SIP seeding with MYSQL_FRIENDS config (sip.conf is coded via a MySQL table) ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Réf.: Re: [Asterisk-Users] Asterisk with UUI support ?
right, so far, here is what I've done:
I succeed in take in a new variable the UUS1 field sent with the connection
request for incoming calls. It was quite simple afterall... (I just had to
find where the data CMSG->Useruserdata is coming in chan_capi.c)
Now I would like to know where this field is instanciated for outgoing
calls in order to control this step? I am looking for that but I don't
really know in which file I need to search...(apparently, it is not in
chan_capi.c)
Can anyone help me?
>Hi Again,
>
>[EMAIL PROTECTED] wrote:
>
>>Can you put this patch on line? (I don't think it's too big...)
>>
>>
>Sorry, i simply don't have a patch, IIRC all i did was inserting
>something like
>
>693a694
> CONNECT_REQ_USERUSERDATA(&CMSG) = "testing";
>
>When i tested this a few month ago, it resulted in something like
>"<40>esting" in the message, but now i can't even get this to work ;-(.
>As i said, my C knowledge is very limited, it's more like cut&past :-)
>
>>In my mind, the main objective is to create a special field and force
>>its value in chan_capi.c and check wether it goes through asterisk or
>>not... What do you think of that?
>>
>>
>Well, i guess the easyest way would be a (new) variable inside >sterisk,
ie
>${UUS1}, on an incoming call it would contain the message, and on an
>outgoing call it could be set to the message we would like to send.
>
>Sorry i can't be of more help, but if you really want this feature, i
>suggest you
>talk to Klaus-Peter Junghanns <[EMAIL PROTECTED]>, i think this would be
>only a few lines of code.
>
>Regards,
>
>Christoph
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[Asterisk-Users] Réf.: Re: [Asterisk-Users] Asterisk with UUI support ?
Can you put this patch on line? (I don't think it's too big...) In my mind, the main objective is to create a special field and force its value in chan_capi.c and check wether it goes through asterisk or not... What do you think of that? Regards -- > >[EMAIL PROTECTED] wrote: >> OK, so I'll do that... Is there any infos I need to know about chan_sip.c >> (because I suppose it's it that I need to play with)? > >Some stuff is already there. This is a "capi debug" trace where i SEND >UUS1 from a normal ISDN Phone TO an asterisk: > > Controller/PLCI/NCCI= 0x101 > CIPValue= 0x10 > CalledPartyNumber = 555 > CallingPartyNumber = <01 81>5551234 > CalledPartySubaddress = default > CallingPartySubaddress = default > BC = <80 90 a3> > LLC = default > HLC = <91 81> > AdditionalInfo > BChannelinformation= <00 00> > Keypadfacility = default > Useruserdata = <04>message > Facilitydataarray = default > >also have a look at chan_capi.c / USERUSERDATA >My C knowledge is *very* limited, but i could send out something with >some wild patching in chan_capi.c, so it's at least possible... > >> Does anyone know an interesting website where I can find infos about UUI in >> ISDN (with CAPI maybe?) ? > >I guess it's somewhere in ITU Q.931, but i dont have this document ;-( > >I also think this would be a very cool feature (i.e. there's a Simemens PBX >that sends out the callername with UUS1), if i can do something else to help, >please tell me. > > >Regards > >Christoph ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Réf.: Re: [Asterisk-Users] Asterisk with UUI support ?
OK, so I'll do that... Is there any infos I need to know about chan_sip.c (because I suppose it's it that I need to play with)? Does anyone know an interesting website where I can find infos about UUI in ISDN (with CAPI maybe?) ? Thanks for your help. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with UUI support ?
Hi there, Is it possible to manage UUI with asterisk and ISDN (T0 Fritz card). Basically, is it possible to send User to User Information using the D-channel, while making a call? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP IAX2 MySQL Config
I've configured asterisk to connect a MySQL database for CDR, Voicemail and SIP/IAX2 peers. - CDR are reccorded - Voicemail config is readen directly in the database but SIP/IAX2 peers still have to be declared in sip/iax2.conf to make calls... However, when I restart Asterisk: [chan_sip.so] => (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found == SIP Listening on 192.168.0.10:5060 == Using TOS bits 0 Connected to database 'asterisk_config' on 'localhost' as 'asterisk_user' == Registered channel type 'SIP' (Session Initiation Protocol (SIP)) == Registered application 'SIPDtmfMode' [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2)) Apr 16 14:10:34 WARNING[1074449120]: chan_iax2.c:6218 load_module: Unable to open IAX timing interface: No such device == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found Connected to database 'asterisk_config' on 'localhost' as 'asterisk_user' == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) == Using TOS bits 0 == IAX Ready and Listening on 192.168.0.10 port 4569 What is the problem? Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Réf.: Re: [Asterisk-Users] Re: External access to voicemail
I am interested too so I think it would a good thing to post an URL (as you said) Thank you![EMAIL PROTECTED])fjåËbú?jË^®+$ºÇ«±:5%H$HJ+ºZµê)¶*'²ø¬Øm¶ÿ+-±Ø é¢oæj)fjåËbú?jË^®+$ºÇ«
[Asterisk-Users] Réf. : [Asterisk-Users] RE: [Asterisk-Users] Réf. : Re: [Asterisk-Users] Fritz ISDN PCI v2 and CAPI
Finally, i will get back to a RedHat 9 distrib as I see that it works with that distribution...[EMAIL PROTECTED])fjåËbú?jË^®+$ºÇ«±:5%H$HJ+ºZµê)¶*'²ø¬Øm¶ÿ+-±Ø é¢oæj)fjåËbú?jË^®+$ºÇ«
[Asterisk-Users] Réf. : Re: [Asterisk-Users] Fritz ISDN PCI v2 and CAPI
I tried that but it still doesn't work... I think I don't have the correct approach. Have I to install any ISDN drivers (=modules ?) BEFORE dealing with CAPI ? If yes, why shouldn't I use the hisax drivers (which are kernel ones) instead of fcpci drivers (which doesn't seems to work, by the way...) And finally, how is it possible to link the two modules together? As you can see, I am drowning in these difficulties and a help would be very welcomed! Cheers! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Réf. : Re: [Asterisk-Users] Fritz ISDN PCI v2 and CAPI
If I do that, the modprobe doesn't find any module called fcpci and is looking for any module called hisax_fcpcipnp as it's the one install for the isdn fritz card. # capiinit modprobe: Can't locate module fcpci ERROR: failed to load driver fcpci Maybe, I should reinstall the fritz card drivers itselves? (however, I don't think so...) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fritz ISDN PCI v2 and CAPI
Hi, First, here is my config: Kernel version 2.4.25 on a Fedora distro, Asterisk and a Fritz! Isdn PCI Card (v2). I try to make the CAPI drivers deal with Asterisk but I can't try to figure out to get of this issue. As I see, Fritz modules are integrated with the kernel, so I directly loaded the 'hisax_fcpcipnp' module from it. I install also Capi modules by downloading archives of the web (make config - make install - insmod...). When i check loaded modules: # lsmod Module Size Used byNot tainted capi 20864 0 capifs 5424 1 [capi] kernelcapi 34048 1 [capi] capiutil 24864 0 [kernelcapi] i810_audio 25244 1 (autoclean) ac97_codec 17236 0 (autoclean) [i810_audio] e100 55496 1 hisax_fcpcipnp 7968 0 hisax_isac 7800 0 [hisax_fcpcipnp] hisax 131820 0 [hisax_fcpcipnp hisax_isac] isdn 139744 0 [hisax] slhc6740 0 [isdn] mousedev5524 0 (unused) keybdev 3140 0 (unused) input 5728 0 [mousedev keybdev] ext3 72896 2 jbd61996 2 [ext3] I've edited /etc/capi.conf and write 'hisax_fcpcipnp - - - - - -' in it. Anyway, 'capiinit' failed (and so for 'capiinfo'): # capiinit ERROR: failed to load driver hisax_fcpcipnp # capiinfo capi not installed - No such device or address (6) Please, could you explain me how to make CAPI drivers speak with the hisax_fcpcipnp module ? Cheers. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Réf. : [Asterisk-Users] IAX2 transfers - it's great!!!!
I am interested in such a conf! IAX2 seems to be a very reliable protocole. Could you please show how to do call routing? Beside, is it possible to do register preferences with IAX2 (or SIP afterall) between an IP phone and proxies? Cheers for consideration. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 183 SIP status with vegastream ip gateway
Hello, I've got problems to configure ip vegatsream gateway products with Asterisk. I've got a Vega50 FXS ip gateway to connect POTS to IP phones but when i pass a call, even if the correct port is 'responding' (the LED is on on the vega50 interface), POTS doesn't ring... The sip registration is ok but the call stops at the 183 sip status... Beside, i try to configure a ip FXO Gateway (vega50 FXO BRI) to connect the ISDN. The problem is the same: the call can't go through the sip status 183... I haven't found any support on the web so please help me! sip.conf: [0101];vega50 FXS sip config type=friend host=dynamic defaultip=192.168.0.10 dtmfmode=inband mailbox=0101 context=default [vega50] ;vega50 FXO sip config type=friend username=vega50 password=password host=192.168.0.11 dtmfmode=inband - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Réf. : RE: [Asterisk-Users] chan_sip.c Error
Well, it's an IP phone (PingTel) but what i did to resolve the pb is that I comment the lines: maxexpirey=180 defaultexpirey=160 in the 'general' tag of sip.conf file. Now, I haven't got this error anymore and the communication is OK... It sounds good like this!Ë^®+$RÇ«²f¢)à+-Ë^®+$RÇ«²X¬¶Çb+¦r¡¶ÚþX¬¶Çb+¦r¿¨¥©ÿ+-wèý«-z¸¬ë ®ÄèPÔ (®ê]jר¦Ø¨Ëâ²+a¶ÚþX¬¶Çb+¦r¿¨¥©ÿ+-wèý«-z¸¬ë ®
[Asterisk-Users] Asterisk GUI ?
Hello,I am just starting to use the Asterisk Solution and I am looking for a GUI which can expose easily all Asterisk possibilities. I tried many things such as asterisk web management, astguiclient, astweb and even gastman but what I would like to have is a 'soft' which propose possibilities in order to assimilate every aspects of Asterisk. Does anyone know something about it?Cheers. ,µêâ²E,z»&j)b b²Ð,µêâ²E,z»%Ëlv("ºg(m§ÿåËlv("ºg(ùYùb²Ø§~ڲ׫É.±êìN
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[Asterisk-Users] chan_sip.c Error
Hello, I have implemented Asterisk on a private network (192.168.X.X/24). I've configured SIP peers and when I make a call between them, I've got these error messages in the * CLI and the communication stops by itself after 30 sec. WARNING[1142106560]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 103 (request)WARNING[1142106560]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 104 (request) Anyone can tell me why? I haven't found real solutions on the mails archive... Thanks you! -In sip.conf:[1982]type=friendusername=user1mailbox=1000host=192.168.111.2context=default In extensions.conf...exten => 1982,1,Macro(std,SIP/1982)...-Ë^®+$RÇ«²f¢)à+-Ë^®+$RÇ«²X¬¶Çb+¦r¡¶ÚþX¬¶Çb+¦r¿¨¥©ÿ+-wèý«-z¸¬ë ®ÄèPÔ (®ê]jר¦Ø¨Ëâ²+a¶ÚþX¬¶Çb+¦r¿¨¥©ÿ+-wèý«-z¸¬ë ®
