similar to: Handling SIP 404 event

The problem is some calls from the PSTN have hidden caller id so if you want to change it to something else then modify chan_sip.c #define CALLERID_UNKNOWN "Asterisk" I've changed mine to: #define CALLERID_UNKNOWN "Unknown" -----Original Message----- From: Shaun Ewing [mailto:sewing@gmail.com] Sent: 22 September 2004 14:16 To: Asterisk Users Mailing List

Typo in your OS79XX.TXT P00 ? instead of P0S !? -----Original Message----- From: Michael L?jtnant [mailto:ml@zyxel.dk] Sent: 17 August 2004 13:31 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Cisco 7.2 firmware for SIP 7940/7960 released Hi Shaun, Saw you post, and rushed to their ftp-server and downloaded it :-) But, I can't make my phone (7940) upgrade, so maybe you

Hi All, Just a heads up - I was looking around the Cisco FTP a little while ago and noticed that the SIP 7.2 images for Cisco IP Phone 7940/7960 were released yesterday (16th August). No new features - all bug fixes according to the release notes. I've already started using it. I thought those of you running the Cisco phones and the appropriate access who didn't yet know would like to

Hello all, I currently have an Eicon Diva Client isdn card using i4l. Outbound dtmf doesn't work (and never has), but there has been an annoying problem with false dtmf detection in calls (that could be triggered easily by blowing into the receiver on the remote end). I looked through the list and found two patches that need to be applied - 1 to isdn_tty.c in the kernel, and another to

help me in another issue related also to registering asterisk with another softswitch: A) If nat=yes, then I have to set canreinvite=no to be able to register, correct? B) In case of using firefly softphone, how it possible to set it to have nat=yes (at the firefly it self and not at the sip user configuration section)? As most of the sip endpoint give an option to set nat=yes and so on, how it

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

Hi All, I want to use Asterisk for VoiceMail for a softswitch. I can dial in to leave voicemail and retrieve. Now there are many SIP Endpoints registered to the Softswitch. The Asterisk is sending a NOTIFY msg to the Softswitch on <ip addr>:0 Somehow Asterisk Looses the port from where the INVITE came in, this NOTIFY msg is not going out of the Asterisk, I cannot see in Ethereal.

Hello all, Over the weekend, I setup and linked an Asterisk box at another site to the Asterisk box here. The phones here are a mixture of Cisco 7940/7960 and Grandstream BT-100 phones. The phones at the other end are Grandstream BT-100 SIP phones. The Cisco phones run SIP 7.1 (upgraded last Friday from 6.1), the Grandstream phones run 1.0.4.68. Both Asterisk boxes are running stable CVS

Thanks Brian, and thanks again for the included definitions <grin> - that helped too. Your comments are really helping clear many questions. I suppose our intensions are to become an IXC. So if my local carrier is sporting old technology, they'll provide TDM services. So if I understood you correctly, the "in-band signaling" is typically SS7, and the alternative is typically

I have a Cisco 7940 when you call in from outside and dial the Cisco phone extension I get this Read_channel ## vpb/1-3: Setting record mode, bridge = 0 WARNING[18451]: File chan_sip.c, Line 1111 (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)

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

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

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, I've got a problem with connection to my SIP provider. In general, everything works, but I get lots of these messages: Oct 17 19:10:06 DEBUG[29707]: chan_sip.c:11148 handle_request: That's odd... Got a response on a call we dont know about. Cseq 42710 Cmd SIP/2.0 Oct 17 19:10:06 DEBUG[29707]: chan_sip.c:11148 handle_request: That's odd... Got a response on a call

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

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:

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

I've noticed that a few people here are from Australia. I'm wondering where you all get your hardware from. I'll probably order some of the X100P cards from Digium soon (and possibly some FXO cards), but for other things such as ATAs, etc. However - these obviously aren't ACA compliant. Does anybody know of compatible hardware (for POTS lines) that is ACA compliant? The cheapest

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, I use in Production : Asterisk 1.2.9.1 We Use Asterisk as a SIP Transit Server to record centrally all the calls. The call flow would be: incoming calls : PSTN -> GW -SIP-> Asterisk(Record) -SIP-> Softswitch -> IP Phone outgoing calls : IP Phone -> Softswitch -SIP-> Asterisk(Record) -SIP-> GW -> PSTN Dial plan in Asterisk is quite simple: [record] exten =>