Re: [asterisk-users] Codec by Network?

you can change the codec on the fly using ${SIP_CODEC}. -- thanks, Yusuf ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Understanding RTCP in Asterisk

X-ECN Telecoms-MailScanner-Information: Please contact ECN Telecoms for more information X-ECN Telecoms-MailScanner: Found to be clean X-ECN Telecoms-MailScanner-From: [EMAIL PROTECTED] X-Spam-Status: No My third try, humph! Yusuf wrote: > Hi, > > I am trying to understand the RTCP

Re: [asterisk-users] Failover SIP logic

But its very sequential, i.e. will try trunk1, then trunk2, then trunk3. If you want to replicate round-robin, r, then do this: [globals] IPt=trunk1-trunk2-trunk3 COUNTt=0 NoOfChannels=3 [just-an-idea] exten => _X.,1,Gotoif($["${COUNTt}" = "${NoOfChannels}&q

Re: [asterisk-users] Upgrade Procedure

r don't. Please check out the default configs first, look in asterisk-1.4.8/configs/ -- thanks, Yusuf ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] How many number of parallel calls can make through asterisk

alls can we make > through asterisk?? > > Regards, -- thanks, Yusuf ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] different codec for different extensions

ion the codec will be G711 and >>> when user call IVR the codec must be GSM >>> >>> >>> Please help me >>> >>> >>> Thanks >>> >>> Nasir Iqbal >>> >>> >>> >>> _

Re: [asterisk-users] Error: Unable to allocate RTCP socket: Too manyopen files

This was a bug 1.4.4 It has now been fixed in Asterisk 1.4.5 Stuart Bennett wrote: > Hi Yusuf > > A friend of mine had the same problem with a high volume site.. The problem > lies with a limitation in Linux. Linux will only allow a certain amount of > open files at a time. You w

[asterisk-users] Error: Unable to allocate RTCP socket: Too many open files

Hi, I have a Intel Xeon Dual Core server, with 3 GB RAM, running Centos 5.0, Asterisk 1.4.4 and mysql 5.0. It is a kinda high-traffic box, with about 60 concurrent calls. The profile of calls on this box are: Incoming: via a Sangoma A101 via SIP from anothjer SIP server Outgoing all calls tha

Re: [asterisk-users] multiple host= in sip.conf

Thing is, he does not REGISTER to me, he just uses me as proxy for his calls. I authenticate his calls in his IP. Alexandre VERNIOL wrote: Not supported jsut use host=dynamic with username and secret. Alex Yusuf a écrit : Hi, I am running Asterisk 1.4.4, and needed to setup sip accounts

[asterisk-users] multiple host= in sip.conf

so I can authenticate him based on his IP in just one place? -- thanks, Yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/ast

Re: [asterisk-users] hanguponpolarityswitch - where did it go??

guring this variable, to change it you might need a restart. So hanguponpolarityswitch only gets looked at on startup, not reloads. -- thanks, Yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRI

Re: [asterisk-users] Digium B410P Need Help

face port (for phones) -> Layer 2 protocol 0x0202 is detected, but not allowed for NT lib. * Port NOT useable for PBX Hi, in /etc/misdn-init.conf, switch the mode to te_ptmp= or something. -- thanks, Yusuf ___ --Bandwidth and Colocation provided by E

[asterisk-users] CDR and RADIUS (cdr_radius) - working

using google, even voip-info has nothing on this module? -- thanks, Yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk with Radius users authentication

t(TIMEOUT(absolute)=${h323-credit-time:17}) ;exten => _X.,10, AbsoluteTimeout(${h323-credit-time}) exten => _X.,10,Goto(sip-calls,${EXTEN},1) exten => _X.,11,Hangup exten => T,1,NoOp(timeout) -- thanks, Yusuf ___ --Bandwidth and Coloca

[asterisk-users] No Ringback, only on 1 SIP provider

Asterisk2, I get the 'making progress passing it to xxx', but I dont hear ringing, then the person answers. -- thanks, Yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update op

Re: [asterisk-users] To jitter buffer or not to jitter buffer?

t using JB on IAX with trunking seems to cause a few problems. -- thanks, Yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] TDM400 with 1 FXO

a reload these are ingnored. I think a 'stop now' would get these settings. -- thanks, Yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.di

Re: [asterisk-users] New user question (X100P)

e seen this also, only on the TDM2400. I think it might be because it, i.e. this cards, takes a bit longer than other cards to initialise, then when ztcfg is run, the card is not ready yet. So I too (hangs head in shame), put something in rc.local t

Re: [asterisk-users] Help with semaphores

nyone can offer. Mitch Thompson Hi, dont know if this is what you looking for but, there is something called macroexclusive, new in 1.4, written by Steve Davies. Read the file in asterisk-1.4.0/docs. HTH -- thanks, Yusuf ___ --Bandwidth and Colocat

Re: [asterisk-users] Dialplan programming vs. AGI vs. ???

entirely when the situation requires it. Personally, I'm glad that there is so many different ways to interact with Asterisk. Nice having a swiss army knife ;) Could'nt have said it better! -- thanks, Yusuf ___ --Bandwidth and Colocation

Re: [asterisk-users] Dialplan programming vs. AGI vs. ???

u We have chosen to do certain funtions in AGI using PHP because we do connections to mysql and some other stuff, and and I think you have much more control with DB-related issues with AGI then with normal dialplan(.conf or AEL). However

[asterisk-users] Re: [asterisk-dev] Dynamically Adding A Context

t an error message instead of it creating the context for me. Any method will do, AGI, AMI, CLI... I just need a solution :) This is a users question. Moved there What about using Realtime??? -- thanks, Yusuf ___ --Bandwidth and Colocation provided by Ea

[asterisk-users] Comments on Billing reconcillation with providers

Hi, I just want out find out how to do bill recon's when you send calls to a provider. They send me their CDR's, and when I compare it to my * CDR's, some calls are 1 second off, either way. How in general is it done by others? -

[asterisk-users] NewTopic - Asterisk and Cisco AS5300 via E1/PRI

call alias exec active show call activ voice br ! line con 0 exec-timeout 0 0 line aux 0 line vty 0 4 exec-timeout 45 0 password 7 ### ! end -- thanks, Yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This messag

[asterisk-users] Operate on registrations

can start there. Any Ideas on how I can get something like this? -- thanks, Yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to

Re: [asterisk-users] Cisco AS5300

Andrew Pogrebennyk wrote: Hello Yusuf yusuf wrote: Hi all, I realize this is OT. I just got a Cisco AS5300, and I need to configure it like such: Asterisk -(H323/SIP)--> Cisco - (E1/PRI)--->Telco So calls originate from the Asterisk side (registered users on SIP or

Re: [asterisk-users] Outbound IVR for Asterisk

, neither commercial apps. Actually, you 100% right, call files with the correct target in dialplan will do it. thanks, Yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous co

Re: [asterisk-users] function call out of AGI script

call variables, then you do your own thing, hit a DB, etc... -- thanks, Yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be

Re: [asterisk-users] How to detect long calls

. Hi , similiar thing happend to me. Try looking at the L() optin in Dial. I define a max call time, say few hours, then warn every x seconds, then cut the call. -- thanks, Yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean

Re: [asterisk-users] MFC/R2 problems

. Greets On 1/8/07, yusuf <[EMAIL PROTECTED]> wrote: Hi, if that means I should post my config, here goes: zaptel: span=1,1,3,cas,hdb3,crc4 cas=1-15:1101 cas=17-31:1101 unicall.conf: protocolvariant=id,10,10 protocolend=cpe group=1 channel => 1-15 channel => 17-31 wanpipe1.c

Re: [asterisk-users] AGI - Getting the passed parameters

e doesn't expose it as part of the std variables mike Hi, in your AGI use: GET VARIABLE CODE thanks, Yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content

Re: [asterisk-users] MFC/R2 problems + Orion GSM Gateway

> [1/ 1000/Clear fwd /Seize ack] The below output(in the mail) is of an outgoing call from Asterisk. Can anyone please help me to see what is wrong? yusuf wrote: Hi, if that means I should post my config, here goes: zaptel: span=1,1,3,cas,hdb3,crc4 cas=1-15:1101 cas=17-31:1101

Re: [asterisk-users] MFC/R2 problems

[w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = NO Josué Conti wrote: Hi Yusuf, how are you? It orders in the list its configurations, so that let us can help. Best Regards Josue 2007/1/8, yusuf < [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>>: Hi all,

[asterisk-users] MFC/R2 problems

UniCall/1 1001 -> [1/ 1/Idle /Idle ] -- Hungup 'UniCall/1-1' What does "- Unicall/1 protocol error. Cause 32769" mean, and can anyone help me. -- thanks, Yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is b

Re: [asterisk-users] jitterbuffer on sip.conf

[EMAIL PROTECTED] wrote: In iax.conf there is option jitterbuffer how about sip protocol ? Are jitterbuffer can configure in sip.conf ? Thanks, for your share If you upgrade to 1.4, there is a jitterbuffer available now for the SIP channel. -- thanks, Yusuf -- This message has been

[asterisk-users] Cisco AS5300

go out on the E1. Cisco is running IOS 12.1.5-12.2.13a I realize this is alot of questions, so please bear with me :) -- thanks, Yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for virus

Re: [asterisk-users] Calls disconnected after 1 hour

at a certain time (not that I know how to do that). Well, it could be that you have the L() option in your Dial string. Or also, if your I connected to a PBX via E1 or something, they usually cut calls at 1 hour. -- thanks, Yusuf -- This message has been scanned for viruses and dangerous

Re: [asterisk-users] Asterisk + Orion E1 GSM Gateway

=asreceived channel => 1-15,17-31 I just cant get the E1 sync light on the Orion to light up green(according to the manual) I have tried crc on/off, pri_cpe/pri_net. I'm kinda running out of ideas! :) Lex Lethol wrote: Hi yusuf, I am working right now on a similar setup. If its

Re: [asterisk-users] AGI Help Please

onnected to the first Ast terminal. So start Asterisk like 'asterisk -cvvvvvv', then you will see output from your AGI. -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. __

Re: [asterisk-users] Asterisk + Orion E1 GSM Gateway

Leo Ann Boon wrote: yusuf wrote: Hi, I just got hold on an Orion E1 30 port GSM Gateway, and I am having problems trying to get the E1 link to come up. I am using Asteisk 1.2.12 with a Sangoma A101 card. I am quite familiar with E1's, both the Digium and Samgoma types, as I

[asterisk-users] Asterisk + Orion E1 GSM Gateway

s and such, but I just cant seem to get this one to work. None of the 30 channels 'come up'. What signailling, crc checking, should I be Master or slave? Anybody have experience on this? -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScan

Re: [asterisk-users] AGI interaction with php

, and it seems to work fine. So go for it! -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users ma

[asterisk-users] Asterisk: SIP Gateway or Proxy

Hi, I realise this might be an insane noob question, but I'm on a huge brain freeze, and I'm trying to decide this: Is Asterisk a SIP Gateway or SIP proxy? -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed t

Re: [asterisk-users] Pickup *8 with CallerID

zapata.conf userincomingcalledidonzaptransfer=yes -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users

[asterisk-users] Ast 1.4 and B410p

I have used the b410p card with Asterisk 1.12 quite successfully. I now want to get the card to work with Asterisk 1.4.0beta3. It however can't seem to get chan_misdn compiled. In menuselect, chan_misdn has this: Depends on: isdnnet, misdn, suppserv Can anyone help? -- thanks,

Re: [asterisk-users] Sangoma A101 gives 'no PRI configured on span 1' error

OK 0 0 0 -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing l

Re: [asterisk-users] SRTP

Khaled wrote: I installed libsrtp can any one help me how to ingrate it with asterisk .to make SRTP Regards Hi, I dont think SRTP is supported in Asterisk. There is some work to have RTP over TCP, where be default its over UDP. -- thanks, yusuf -- This message has been scanned

Re: [asterisk-users] Zap channel shows "answered" as soon as outbound ringing starts

nd Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be

Re: [asterisk-users] ${CALLERIDNUM}

2,1,GotoIf($["${CALLERIDNUM}" = ""]?5) This will tell it to jump to 5 if callerID if but how do i tell it do jump based on length of callerID? Hi, would this work: exten => _X.,4,GotoIf($[${LEN(${CALLERIDNUM})} != 3 ] ? 40) -- thanks, yusuf -- This me

[asterisk-users] AEL2 - CUT function usage

ion has been removed in 1.4, so now I am usung the CUT function, but where is it explained that you have to have to use SET and the commas ',' has to be replaced with '|'. Or have I done something stupidly wrong :) -- thanks, yusuf -- This message has been scanned fo

Re: [asterisk-users] Asterisk Call Statistics

. Hi, If you have asterisk-addons, you can get all CDR's, which include all the above statistics, written to a MySQL or PGSQL database. It would then be very easy to get this on to a web page. -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner

[asterisk-users] Please help with these SIP errors

ock_read: BAD! BAD! BAD! == Spawn extension (iax, 0825905581, 24) exited non-zero on 'SIP/sipBBG-b736f910' -- thanks, yusuf -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. __

[asterisk-users] Multiple bridge attempts

p/7-1 -- Attempting native bridge of Zap/1-1 and Zap/7-1 -- Attempting native bridge of Zap/1-1 and Zap/7-1 -- Attempting native bridge of Zap/1-1 and Zap/7-1 -- Attempting native bridge of Zap/1-1 and Zap/7-1 -- thanks, yusuf -- This message has been scanned for viruses and dangero

[asterisk-users] Please explain these SIP errors

ock_read: BAD! BAD! BAD! == Spawn extension (iax, 0825905581, 24) exited non-zero on 'SIP/sipBBG-b736f910' -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___

Re: [asterisk-users] Inaccurate CDRs

it where my billsec were wrong for some other reason. Try callprogress=yes in zapata.conf, although I dont even think this will help, but you can try. -- thanks, yusuf -- This message has been scanned for viruses and dangerous conten

Re: [asterisk-users] Student Research - Asterisk H323 Video

be using? Have you tried Ekiga! http://www.gnomemeeting.org/ I supports video also. thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provi

[asterisk-users] 12 port FXx PCI card

Hi, http://www.openvox.com.cn/products_detail.php?genre_id=17&id=45 The A1200P is a 12 port card, that used the same modules as a TDM400P. I have been looking at this card, and I want to know if anybody has used this card and what their experiences were? thanks, yusuf -- This message

Re: [asterisk-users] asterisk-addons-1.2.4 Installation Problem

. Regards Hi, the mysql-devel package needs to be installed, because you need the headers. -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -

Re: [asterisk-users] Unknown signalling method 'pri_cpe'

d_module failed, returning -1 Oct 3 13:04:02 WARNING[5823]: loader.c:554 load_modules: Loading module chan_zap.so failed! Ouch ... error while writing audio data: : Broken pipe I think its because you dont have libpri installed. Install libpri, then try! -- thanks, yusuf -- This mes

Re: [asterisk-users] Asterisk Advice

2GB RAM - this box can be upgraded or changed as well and am running RHEL 4.0 AS on it. Thanks very much Best wishes Iyer Hi, PRI cards come in 1/2/4 ports, so just 1 card will be enough to manage three PRI lines. Also look at Sangoma cards, they have PRI carda also. -- thanks, yusuf

RE: [asterisk-users] Cisco CAll Manger and H323

Thanks Dan, that was awesome, and really made sense about what was really happening. :) Will try a newer. BTW: I did get it to successfully route inbound calls to asterisk with oh323, and DTMF and transfers worked fine. > Yusuf wrote: >> Hi Dan, > >> I used asterisk 1.

Re: [asterisk-users] Cisco CAll Manger and H323

x27;t help, I can try to provide more details. Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yusuf Sent: Thursday, September 28, 2006 10:33 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Cisco CAll Manger and H323 Hi, I recently h

[asterisk-users] Cisco CAll Manger and H323

the call, and int he debug it says: "stopped from reciving frames from OOH323/cisco , bridging is being stopped". What is wrong? What RTP ports must I be using? thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed t

Re: [asterisk-users] new in 1.4?

ocation provided by >> Easynews.com<http://easynews.com/>-- >> > >> > asterisk-users mailing list >> > To UNSUBSCRIBE or update options visit: >> >http://lists.digium.com/mailman/listinfo/asterisk-users >> > >> >&

Re: [asterisk-users] Iax Netstat Output

-- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options vis

Re: [asterisk-users] RTCP and RTP packetization in 1.4

ou will be introducing delay. -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To U

Re: [asterisk-users] RTCP and RTP packetization in 1.4

Matt Riddell (IT) wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 yusuf wrote: Hi all, I'm so excited about 1.4 coming out soon :) , I was wondering if anyone can comment on the following: 1. Will RTP packetization (5162) committed to trunk (43243) be in 1.4? I have it running

[asterisk-users] RTCP and RTP packetization in 1.4

Will RTCP (2863) committed to trunk (32230) be in 1.4? There is only a patch for 1.2.4, have used that, but will there be an updated patch. -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be

Re: [asterisk-users] How to Dial a number with Sangoma PRI card?

le? For example: ChanIsAvail(Zap/1&Zap/2&Zap/3) or ChanIsAvail(Zap/1-1&Zap/1-2&Zap/1-3) And, once discovered which channel is available, which form of Dial should I use? Should I say: Dial(Zap/2/1234) or Dial(Zap/1-2/1234) yusuf wrote: hi, I did it like this: I wrote a P

Re: [asterisk-users] How to Dial a number with Sangoma PRI card?

ling list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation

Re: [asterisk-users] Has anyone tried to install both digital card

hnicians. Go for it. thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBS

[asterisk-users] SOLVED: ringback on box with E1 and premicell

ring when answeronpolarityswitch=yes SOLUTION: in zaptel.conf loadzone=za in zapata.conf callprogress=yes progzone=za priindication=outofband -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed t

Re: [asterisk-users] Satellite link-IAX Jitter Buffer.

ffer (it has PLC). iax.conf says which jitterbuffer settings applies to the 'new' or 'old'. Try the new settings. Here is one I have (the 'new'): Asterisk 1.2.6 ;jitterbuffer=yes ;forcejitterbuffer=yes ;maxjitterbuffer=300 ;maxjitterinterps=300 ;resyncthreshold=1500

Re: [asterisk-users] 0005162: RTP Packetization : Few questions

. Hi, I will try this. But even with autoframing=no, B still sets ptime:20. on B in sip.conf [sipacket] username=sipacket secret=sipacket type=friend host=dynamic context=default disallow=all allow=g729:60 ;autoframing=yes ;canreinvite=no -- thanks, yusuf -- This message has been scanned f

Re: [asterisk-users] 0005162: RTP Packetization : Few questions

r-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing NoOp("SIP/192.168.0.196-09fea900", "YUSUF") in new stack -- Executing Playback("SIP/192.168.0.196-09fea900", "demo-congrats&qu

Re: [asterisk-users] 0005162: RTP Packetization : Few questions

tting packetization to 20, when it should be 80, and is not respecting autoframing. I have tried this with reinvites=yes and no, and autoframing=yes and no, still the same. Also, I am not sure if this is a bug. If in sip.conf, if I set [yusuf] username=yusuf secret=yusuf type=friend callerid=100

Re: [asterisk-users] Asterisk and NAT ?

ou will be able to call/receive on both. -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users ma

Re: [asterisk-users] Zaptel-1.2.8 compile problem

rsion 3.4.5 20051201 (Red Hat 3.4.5-2)) #1 Wed Mar 8 00:07:35 CST 2006 most probably: http://bugs.digium.com/view.php?id=6425 -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. __

[asterisk-users] 0005162: RTP Packetization : Few questions

[general] context=default ; Default context for incoming calls disallow=all; First disallow all codecs allow=ulaw:20 allow=alaw:20 allow=g729:80 autoframing=yes am I doing something wrong? Also, I am not sure if this is a bug. If in sip.conf, if I set [yusuf

Re: [asterisk-users] valgrind + Asterisk

Hi, Tzafrir Cohen wrote: On Thu, Aug 17, 2006 at 02:37:52PM +0200, yusuf wrote: Hi, has anybody got valgring to work with asterisk Yes i do a -- valgrind --tool=memcheck -v asterisk -c then Asterisk just dies. What version of asterisk? Did you use any special build options

[asterisk-users] valgrind + Asterisk

. So Asterisk might have a memory leak, and I am trying to find it. Can anybody help? thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation

Re: [asterisk-users] Asterisk Real Time and sip.conf file used at

Yes, you can use both at the same time. The only restriction is that > you cannot use the realtime static configuration and realtime > configuration. > > -- > Hi realtime static configuration and realtime configuration??? What is the difference, can you please explain? thanks,

Re: [asterisk-users] Asterisk suddenly die

ose then restart asterisk, if it dies again, you can check /var/log/asterisk/full for info -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocat

Re: [asterisk-users] Voismart GSM - no billsecs

Thanks to Matteo, everything is going right (also sms in and out) Hi, I used the above devel version of visdn, and now asterisk does not even pick up the sims. can you please tell me what linux flavor you using, kernel version, Asterisk version can you also tell what udev veriosn you ha

[asterisk-users] Voismart GSM - no billsecs

Hi all, I have a Voismart GSM card. I have calls through going fine. But in the cdrs, all the calls have disposiotion of "NO ANSWER" and the billsecs are 0. I am using Asterisk 1.2.7, visdn 0.16, kernel 2.6.11-12, on CentOs 4.2 can anyone help? -- thanks, yusuf -- This messag

[asterisk-users] [Fwd: [Fwd: polarityswitch: no ringback]]

talk. So therefore I do not hear the phone ring when answeronpolarityswitch=yes Can anyone help? -- thanks, yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

[asterisk-users] FXS: No ringtone

one. Yet other calls from the PBX, non cell calls, have ringtone. So when a call uses the E1 anf FXO, I get no ringtone. Has anyone seen this before -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed t

[asterisk-users] AGI: Channel status

TUS IS:"+trim(fgets(STDIN,100))); what am i doing wrong -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com --

Re: [Asterisk-Users] H323 Asterisk best practices

which can hook up to OpenH323 or Opal. try it! -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users

Re: [Asterisk-Users] h323 with asterisk problem

y different combinations of faststart, h245Tunnelling,h245inSetup. Also, you can try the ooh323 in asterisk-addons, or oh323 from in-access networks. thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. _

Re: [Asterisk-Users] Re: Sangoma A101 configuration

he wanpipe drivers correcly when running ./Setup install yusuf Kamran Ahmad wrote: I have followed these two for configuration of sangoma A101 http://www.ss7box.com/s01_setup.html http://www.ss7box.com/support_wancfg_1.html on my side "wanrouter star/restart" is working fine when

Re: [Asterisk-Users] Limit outgoing calls

GROUP) exten => _X.,2,GotoIf($[${GROUP_COUNT()} > 30 ] ? 4) exten => _X.,4,NoOp(This trunk has more than 30 calls) thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. __

[Asterisk-Users] iax2: dropping too many packets

resyncthreshold=50 trunktimestamps=yes why is soo many packets getting dropped at Branch B. the network traceroutes are same from both sides. Am i doing something wrong in asterisk. Is linux maybe dropping packets. I really appreciate any forthcoming comments/suggestions. thanks, yusuf

[Asterisk-Users] dialing FXO gives wrong billsec

RED', even though the cell phone was not answered. my dial string looks like so: (all calls come in to inbound) [inbound] exten => _X.,1,Dial(ZAP/1) I have a standard zaptel and zapata, Asterisk 1.2.6 thanks, yusuf -- This message has been scanned for viruses and dangerous con

Re: [Asterisk-Users] Error messages

er to automatically load all then specifically noload whichever you dont want with a noload =>, or with autoload=no, specify which you want to load. -- thanks, yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing l

Re: [Asterisk-Users] aterisk+h323 trunk?!

if gatekeeper: Dial(H323/[EMAIL PROTECTED]) depending on what you install your Dial is Dial(H323...) or Dial (OOH323) -- thanks, yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update op

[Asterisk-Users] oh323: asterisk crashes on a dial

h ID ba528228 i can provide the trace file (with log level 10) can anybody provide some feedback/comments -- thanks, yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] Performance: Xeon or Opteron?

sterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update

Re: [Asterisk-Users] correct version of asterisk for oh323

: chan_oh323.so: load_module failed, returning -1 == Cleaning up OpenH323 channel driver. Apr 18 17:47:39 WARNING[11385]: loader.c:554 load_modules: Loading module chan_oh323.so failed! I am using FC3 with 2.6.5-1.358 kernel. Any suggestions? yusuf Herchi Silviu wrote: Hello, I've used Asterisk

[Asterisk-Users] correct version of asterisk for oh323

Hi, i have been using asterisk CVS 19/07/2005 and asterisk-oh323-0.7.2. I now want to use oh323 with Asterisk 1.2.4+. Can anyone tell me what versions of oh323(and pwlib and oh323) they got to work with Asterisk 1.2.4+. -- thanks, yusuf

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