Re: [Asterisk-Users] YAC and IPs
On Tue, 2005-04-26 at 08:58 -0500, Anton Krall wrote: Im using YAC to send callerid info to PCs and I was wondering if there is a way to get the IP of a certain SIP or IAX client/technology when a dial command is issued. For example, if the dialplan has a dial sip/client or iax2/client, is there a way to get the current clients IP so I can pass the parameters to the system call that send the YAC callerid info? I use the same system. I have a couple of perl agi scripts that fill the gap. One of them finds a hardcoded IP in a hash of extensions, the other sends the callerid information in YAC format. Email me if you want a copy. Adam. p.s. CC adam@mydomain to make sure I see it if you reply. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] first few seconds of outgoing calls cut off
On Sat, 2005-04-16 at 13:50 -0700, snacktime wrote: This also happens to me when I call into my own * box voice system unless I'm very careful about adding appropriate wait statements after answering the line. Not sure if this is related to the above problem, but it made me wonder if an * box somewhere in the path of my outgoing calls might be the culprit. Any thoughts? This problem could be caused by a SPF firewall somewhere in the path. I had a similar problem where the firewall was dropping RTP packets in one direction until it saw a packet in the other direction. Removing the stateful firewall rules and replacing them with pairs of non stateful rules fixed the problem. Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Find extension from Dial(,M()) macro
Hi,
How do I find out the extension that answered an incoming call from
inside a Dial(number,M()) macro? I've tried MACRO_EXTEN but it seems to
be empty.
exten = s,100,Dial(${EXTENSIONS},40,M(support))
[macro-support]
exten = s,1,NoOp(${MACRO_EXTEN})
etc...
Thanks in Advance,
Adam.
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Re: [Asterisk-Users] Find extension from Dial(,M()) macro
On Wed, 2004-11-24 at 10:34, Adam Greenbaum wrote:
How do I find out the extension that answered an incoming call from
inside a Dial(number,M()) macro? I've tried MACRO_EXTEN but it seems to
be empty.
Always the way isn't it, discovering the answer just after you post!
I can find the extension by chopping the end off the ${CHANNEL}
variable.
Adam.
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[Asterisk-Users] Re: List proposition
On Tue, 2004-11-23 at 20:26, [EMAIL PROTECTED] wrote: Hello everyone! We have been thinking about somethingearly snip With respect, this could have come from anyone... and looks like it did from the DNS records. I appreciate it's not an easy situation, but there must be some way of doing this where it does not look like a scam? Tell tale signs: - Don't tell anyone about this - Anonymous receipt of funds - Newly registered domain Anyone else agree with me on this... or actually get the original mail? Adam. ___ 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
Re: [Asterisk-Users] IVR and voice mail using G729
On Wed, 2004-11-17 at 18:32, Alvaro Gonzalez wrote: I need to know if it is possible to use the IVR and Voicemail using G729, I have two SIP phones that uses G729 and I can not heard the IVR and the voice mail. Yes, you just need to purchase a G.729 licence from Digium. 2 phones, $20 (Presuming 1 channel at a time). Not a bad deal: http://www.digium.com/index.php?menu=asterisk_g729 ___ 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] Dial without bridge
Hi, Is it possible to dial a number (from an extension dial plan) and not bridge the call immediately? I want to announce something to the callee before the caller is connected. Thanks, Adam. ___ 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
Re: [Asterisk-Users] Dial without bridge
On Fri, 2004-11-12 at 10:34, Selim wrote: You can try a Background command in your dialplan before the Dial statement: Thanks for the reply. Doesn't the background command play to the caller though? I need to send audio to the callee as soon as then pickup, but before the call is bridged. ___ 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
RE: [Asterisk-Users] Dial without bridge
I want to announce something to the callee before the caller is connected. The A(x) parameter of the dial command allows this - see http://www.voip-info.org/wiki-Asterisk+cmd+dial. The caller continues to hear ringing while the callee hears the audio. Once the audio is finished, the call is bridged. You could also use the M(x) parameter to run a macro instead. Excellent, just what I was looking for, sorry for not finding it myself in the first place! 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
Re: [Asterisk-Users] SIP via Wireless Ethernet Bridge and Double NAT
On Mon, 2004-11-01 at 15:37, Paul Rodan wrote: I keep seeing sip registration failed requests on Asterisk. I checked and double checked the passwords, its fine. I believe its that the device gets the UDP packets through to the Asterisk server fine, with the authentication information or whatever; but when the Asterisk server tries to respond via UDP, it doesnt make it through. So it fails. I've seen this situation, it's probably slightly different to how you describe, although has the same effect. AFAIK when you send the register, the asterisk sends back a hash so you can encode the password. If the client doesn't get the reply it will just try again without the password. You need to see what's going on at the ATA end of the network to see if the packets are coming back. Can you tcpdump at any point along the way? If not then set the 'nprintf' host on your ATA and see if that sends any useful info about what is happening. If you don't have the nprintf server software from cisco then you can just use netcat: nc -l -u -p 9001 It sounds as though it'll take you more than 2 weeks to fix, might as well wait for the SDSL ;) Adam. ___ 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.conf registration
Sorry to be a pain reposting this question, but a mail server problem resulted in my last message missing the boat slightly: How do I associate a SIP entity with a registered account on a PSTN gateway? I have 2 register lines and 2 entities in sip.conf. When I dial into asterisk from the PSTN gateway it always associates with the second entity. (CVS) I've looked through the source and it _seems_ as though you can only match against the from: username [find_user()] and from source address [find_peer()]. Surely you would need to match against the destination sip: username, not the From: username. Am I missing something? I must be, otherwise you would never be able to use multiple accounts on a SIP gateway. Thanks for your help, Adam. ___ 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
Re: [Asterisk-Users] Multiple SIP gateway accounts
On Wed, 2004-10-27 at 15:58, Adam Greenbaum wrote: If you have multiple accounts on the same SIP-PSTN gateway, how do you dial out of a particular one? I think the answer will also involve me setting my domain and username on the outgoing invite, but I have a feeling this might not work because of the authentication. Ok, to answer my own question, it looks as though the correct way of doing this is to use [EMAIL PROTECTED] (from the sip.conf) in address you are dialing. Then fromuser and fromdomain in the SIP entity. Would anyone comment whether this is correct? This now brings me onto another question: How do I associate a SIP entity with a registered account on a PSTN gateway? I have 2 register lines and 2 entities. When I dial into asterisk from the PSTN gateway it always associates with the second entity. (CVS) I've looked through the source and it _seems_ as though you can only match against the from: username [find_user()] and from source address [find_peer()]. Surely you would need to match against the destination sip: username, not the From: username. Am I missing something? I must be, otherwise you would never be able to use multiple accounts on a SIP gateway. Thanks for your help, Adam. ___ 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
Re: [Asterisk-Users] Automatic code selection
On Thu, 2004-10-28 at 23:19, Ulrich Holeschak wrote: But if i have i have a connection between the local SIP phones i want to force SIPphone1 to G729A and SIPphone2 to ALAW. Surely you would want alaw - alaw for internal calls? if not then G729-G729, then asterisk would just pass through without the need to do any transcoding. That is unless you are talking about CPU load on the phone. ___ 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] Multiple SIP gateway accounts
Hi, I've searched the archive but can't seem to find the answer to my problem (well, I might have, but 'EMAIL PROTECTED' was taking out some key ingredients). If you have multiple accounts on the same SIP-PSTN gateway, how do you dial out of a particular one? I think the answer will also involve me setting my domain and username on the outgoing invite, but I have a feeling this might not work because of the authentication. Currently My sent invite contains this: From: Adam Greenbaum sip:[EMAIL PROTECTED] and I get the feeling it should look more like this. From: Adam Greenbaum sip:[EMAIL PROTECTED] Thanks for your help, Adam. ___ 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
Re: [Asterisk-Users] ATT Cordless VOIP Phone?
On Wed, 2004-10-27 at 19:42, Me wrote: Anyone know if this could work with Asterisk? In order to make internet calls, you will need a PC running Windows 95, 98, or equivalent; a sound card; and an internet connection that supports VoIP and VoIP client software. So, yes and no but mainly no. ___ 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] G.726
Hi, I am currently evaluating asterisk, I've upgraded to -head to get G.726 support. I seem to get a segfault when placing a call between G.729 (open source) and G.726. Is this expected? Apart from 'don't do that' what are my options? Thanks, Adam. ___ 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
RE: [Asterisk-Users] G.726
On Tue, 2004-10-26 at 17:29, Brian West wrote: G729 works fine here It works here too, as long as I don't try any transcoding! I've tried G.726 and *law, I get a segfault on both. ___ 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
