similar to: Extra parameters in SIP URIs

Hello: I would like to use Asterisk as a redirect/proxy sip server to route SIP calls on a sip header/parameter basis. I've tried some things successfully: - SIP registration from clients. - On-the-fly compression for wan VoIP transfers: SIP G.711 --> GSM IAX --> (wan) --> GSM IAX --> SIP G.711 - Sending custom parameters in URI: exten => 1,1,Setvar,VXML_URL=var1=value1

Dear all, How do I set the UUI informations for outgoing calls and read the UUI information for incoming call in asterisk? Thanks in advance.. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100531/ffcceeee/attachment.htm

Hi, If the special handling in the meg2reg pass is to look for lifetime intrinsics, shouldn't it cast to <IntrisicInst> and then use getInstrinsicID to check for lifetime_start and lifetime_end ? The thing that I don't understand is the following piece of code, which finds all the users and cast it to <Instruction> then eraseFromParent(). How can this guarantee that it only

Hello, I'm trying to registrate a Nuance Server in Asterisk (using SIP) with no success. It seems that Nuance does not send any secret/password (there is no way to define it!), this is the list of parameters that Nuance provides for registration: audio.sip.UserAgentURI=sip:user@domain audio.sip.UserAgentPort=<port> audio.sip.ProxyServerURI=sip:<IP>:<port>

----- Original Message ----- > From: "guoqing zhang" <gqzhang81 at gmail.com> > To: llvmdev at cs.uiuc.edu > Sent: Friday, January 30, 2015 4:29:16 AM > Subject: [LLVMdev] About user of bitcast/GEP instruction > > Hi, > > > In PromoteMemoryToRegister.cpp, it seems to rely on the fact that the > only users of bitcast/GEP instruction are lifetime

I have a similar setup to you and get the same message regularly. I don't think it's the cause of your problem. I did some research on it a while ago: IIRC the cisco uses codec 13 for "silence suppression" whereas asterisk (correctly) uses codec 19. The router can be configured to use 19 also, but I didn't bother. I'm sure somebody will correct me if I'm wrong about

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?

Hi, In PromoteMemoryToRegister.cpp, it seems to rely on the fact that the only users of bitcast/GEP instruction are lifetime intrinsics (llvm.lifetime.start/end). I did some searching in llvm/test folder, it seems to be true. However, by reading LLVM IR manual, I don't see any restriction stated on the possible user of bitcast/GEP instruction. So my question is who impose the restriction ?

Hello, Does asterisk supports VXML? Couldn't find much resource on that on google and wiki. Thanks Foong

Hello, In our SIP network, Asterisk is the central PBX, and it routes calls to the PSTN thru a Cisco Router - IOS 12.2(11)T9. If a client softphone calls directly via Cisco to the PSTN, the call works successfully. If the client softphone calls via Asterisk to other SIP internal extension, it work fine too. The problem is when a client calls an Asterisk extension, and Asterisk transfers

So, I'm trying to get the Asterisk vxml (from i6net) working. Having no luck with it. My dial plan has: exten => _X.,1,Answer() exten => _X.,n,Wait(1) exten => _X.,n,Vxml(file:///tmp/menu.vxml) The /tmp/menu.vxml file has: <?xml version="1.0"?> <vxml version="1.0"> <form> <block><audio

Is it possible to send CLID NAME on a PRI? The numbers we send out are being received by telco and propagated, but the names we send out are not showing up. Is this a feature in PRI? Do we need to set PRI_NET instead of PRI_CPE? Is this just not possible? Is this a telco config issue? Thanks for your help... I've read voip-info, and various other sources, and search engines, and google...

I was browsing through the web config of a Sipura SPA-841 (Firmware 2.0.13) and noticed a setting marked 'paging' under supplementary services on the Phone settings page on the advanced admin login. Anyone know how it might be used? Could it be like the Snom - exten => 10,1,SetVar(VXML_URL=intercom=true) exten => 10,2,Dial(SIP/testuser) Craig

Hello: I am testing Asterisk with oh323. My question is: can Asterisk route some calls thru a second h323 gateway (a h323 <-> PSTN gw)? - Asterisk ip: 192.168.1.10 - h323<->PSTN gw: 192.168.1.20 I've tried: exten => _9XXXXXXXX,1,Dial(OH323/192.1.1.20) or exten => _9XXXXXXXX,1,Dial(OH323/BYEXTENSION@192.1.1.20) but it does not work at all. If my h323 client

Dear All, I'm integrating avaya commuication manager difinity ver 1.0 with asterisk using B2B E1. following are the details of my H/W, zaptel configs and software installed. Digium TE110p asterisk 1.2.19 cent OS 4.4 zaptel 1.2.18 libpri 1.2.4 etc/zaptel.conf span=1,0,0,cas,hdb3 bchan=1-15,17-31 dchan=16 when i ztcfg -vvv im having this error message and the E1 is not getting up. "cas

Hello! Is there a way to communicate from Asterisk to a TTS server? I've seen festival.conf, but it seems that it works only with Festival server. Thank you.

Hello List, I need to connect with an Avaya PBX (this part is done), and i would like to get and send back User-to-User Information (UUI) with the call. The UUI need because I need to identify the call based on something witch is available on Asterisk and Avaya too. It is possible, or have a better solution? Anybody did it before? Thanks for the help! Zsotya

Hi, I came thru ISDN UUI (User-User Information) protocol which is defined in this RFC - http://www.ietf.org/id/draft-ietf-cuss-sip-uui-17.txt But I don't understand how to use this with Asterisk. Any idea would be much appreciated. Thanks. Gopal. -------------- next part -------------- An HTML attachment was scrubbed... URL:

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.

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 ---------------------- > >jean-marie.goupil@telintrans.fr wrote: >> OK, so I'll do that... Is there any infos I need to know