similar to: Anyway to pass CIC in sip header

Hi, Is there any (easy) way to get Asterisk to include CIC-information in the SIP INVITE? CIC: http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/vvfax_c/callc_c/sip_c/sipc1_c/chapter3.htm#1314580 I need my SIP INVITE to look something like: INVITE sip:5550001;cic=+16789@172.18.202.60:5060 SIP/2.0 I'v tried a couple of different things but can't find anything

Has anyone ever setup Asterisk to pass Feature Group D access while using a CIC code for outbound calls? If so can you please email the configuration you have done? I have tried to get this up and running but with no luck. I have also contacted support and I cant seem to get this going. Thanks in Advance, Jason Miller

Dear List; I am trying to create a link between Asterisk and My softswitch, the link to be SIP Trunk. I did the below configuration and I do not know if any one can help me and advise me to have better configuration to be sure that link is fine. But I do not know how to determine the SIP username to be sent for my softswitch as sometimes the softswitch need to check it. Also, does asterisk

Hello, I'm trying to run smbtorture against another system. I have installed version 4.0.0alpha9 locally. The remote system is registered with ADS as: distinguishedName: CN=bl-uits-cictest,CN=Computers,DC=ads,DC=iu,DC=edu name: bl-uits-cictest dNSHostName: bl-uits-cictest.ads.iu.edu servicePrincipalName: HOST/bl-uits-cictest.ads.iu.edu servicePrincipalName: HOST/BL-UITS-CICTEST The

Dear Alex; Thanks alot for your nice help. This is if I need to let Asterisk register with another softswitch (so I used register =>), what if I need asterisk to send call for the softswitch without register to it (directly)? If I removed the register => then how it will distiguish the IP address in the "host" at the [sip_trunk] is the IP address of the softswitch that need to

Dear all what is softswitch what is difference between asterisk and softswitch ?? regards satish patel --------------------------------- Catch up on fall's hot new shows on Yahoo! TV. Watch previews, get listings, and more! -------------- next part -------------- An HTML attachment was scrubbed... URL:

HI all, I have the following setup running: EP<--->Calling Asterisk<--->Relaying Asterisk<--->Softswitch<---> PSTN The Endpoint EP is registered with the Calling Asterisk. Calls are forwarded from this machine to Relaying Asterisk which in turn forwards it to the Softswitch. In addition, this machine also relays back responses from the Softswitch to the Calling

hello, all of users: sorry, resend it again for clarifying the message. I have implemented cha_ss7 in china. initially, the chan_ss7 can not support the call group. i modify the code. now the problem is that, both sides can hear the ring, but i can not hear the voice from each other. i think the ss7 does not send the voice steam to the destination. in chan_ss7, i added:

Pardon my ignorance, but I was hoping someone could clear up something for me. - For a few POTS lines, digium has a single port card for that, or a T1 card to a channel bank. - For 10 or more lines, digium has a T1 or E1 card for that too based on PRI channels - For 100's to 1000's of lines, I suspect a soft-switch is in order??? A traditional phone company will sell: - POTS lines for

Hello averybody, I'm looking the softswitch in digium website, anyone test the softswitch? Best Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071202/2440f782/attachment.htm

Hi List; If I need to do a trunk between Asterisk and another SIP softswitch (so Asterisk will send a SIP calls to that softswitch), then I have to configure this on the sip.conf file or where exactly? And is it the same when I configure iax trunk? Should I determine the context in this case for this SIP trunk? Regards Bilal

Hi guys. I'm working on a VOIP service provider. We have two customers running asterisk. Customer A and B. When A call to B everything is ok. When B call to A the call ring but sip messages didn't arrive on asterisk A. In my softswitch i see the invite sip message sended to A. When every other numbers(TDM and SIP) call do A everything is ok. Have any issue in asterisk that can resolve

I've been working in the VoIP industry for just a bit over a year now... Mostly taking care of the underlying systems. I've now reached the point where I'm being drawn more and more into the call processing side of things. My background is in computer and "classic" telephony systems (DMS250/MTX, DSC 400, T1, channel banks. telabs analog echo supressor modules and

Hello, ? I'm in a bit of a fix. We have a particular Windows based softswitch which is has its SIP and H323 ports hardcoded to listen on a particular IP address. The problem is that the ISP is having major issues and we can no longer depend on them for service. The softswitch will not listen on any other IP address and this can not be fixed. I was thinking of creating a NAT network wherein we

I'm confused a bit about the following and was hoping to get some answers on this group - What is exactly implied when we say asterisk can connect to a PSTN. Does it mean connecting to the PSTN via PRI/T1/E1? If yes, then I assume asterisk does not need to do any SS7 signaling and all it does (playing the role of a PBX) is to connect to a Class 5 Switch at the CO. Is this a correct statement?

Dear All I am trying to specify the output directory and the directory for the log files (the same) for texi2dvi. The default for my windows computer is C:\Users\...\Documents which I do not want. The help guide: texinputs NULL or a character vector of paths to add to the LaTeX and bibtex input search paths. is a little cryptic for me this morning I have tried texi2dvi(file =

I'm learning how to use tapply. Now I'm having a go at the following code in which dati contains almost 600 lines, Pot - numeric - are the capacities of power plants and SGruppo - text - the corresponding six technologies ("CCC", "CIC","TGC", "CSC","CPC", "TE"). .....................................................

I'm making outbound calls by placing call files in the asterisk outgoing directory. At times, the call would be hung by SS7 without even attempting (due to error in the outgoing number). I get the following on console: -- Attempting call on ss7/9297210213 for s at croom:1 (Retry 1) -- Sent IAM CIC=22 ANI=9134904821 DNI=9297210213 RNI= -- SS7 hangup 'SS7/callserver/22'

Hello friends: I am facing cutoffs randomly when negotiating calls. The PBX dials the destination, the provider (softswitch) receives the request *[1]* and sudenly the PBX hangs up the call* [2]* while the provider is still dialing it, as a consequence the remote peer receives a ghost call. Along the atempt I could see six times a messages regarding NAT isuues *[3]* I hope anyone can give me an

Hello everybody, Im new to the list and also new to asterisk, Im wondering if I could set up asterisk as a softswitch, I guess for what I've been reading that It could be possible but almost all the info and documentation Ive found so far is about asterisk as a PBX, etc. Im willing to set a small voip wholesale traffic bussiness and Im not quite sure asterisk is the right chocie for that.