Re: [Asterisk-Users] ATXFER discussion, what's your opinion ?
Alessio Focardi wrote: Hi, I'm experimenting attended calls tranfers and I'm a little bit confused. SNIP I honestly think that transfers is one thing that Asterisk should improve a LOT to be able to stand up to even the most cheapo taiwanese no-name PBXs, which support attended transfers out of the box. I've had two possible clients refuse an Asterisk installation because attended transfers were unreliable. I honestly didn't know how to explain that a feature available in PBXs for decades was not available or didn't always work. I don't know how the current HEAD is going, but so far, attended transfers weren't available in stable. Regards, Mike ___ 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] Wire Tapping on Asterisk
Hi,
Yes, there is a way. In extensions.conf, you add a macro as:
[macro-record-on]
exten = s,1,AGI(set-timestamp.agi)
exten = s,2,SetVar(CALLFILENAME=${timestamp}-${ARG2}-${ARG1})
exten = s,3,Monitor(wav,${CALLFILENAME},m)
then, when you want to record the call, you use:
exten = s,1,Macro(record-on,NAME_OF_CHANNEL,${CALLERIDNUM})
this will record to a file named for example
20050704-173558-93xxx-IN.wav (number obfuscated)
The set-timestamp.agi is nothing else than
#!/bin/sh
longtime=`date +%Y%m%d-%H%M%S`
echo SET VARIABLE timestamp $longtime
MAKE SURE OF THE LEGALITY OF DOING THIS IN THE PLACE YOU WILL BE DEPLOYING.
Best regards,
Mike
Christoph wrote:
On Thu, 2005-07-14 at 17:00 +0800, Ian Bert Tusil wrote:
I'm new to asterisk. I would like to ask if there's a feature in
asterisk wherein you can monitor ongoing calls, some kinda like
tapping into active phone calls? It must have this feature but I do
not know where to get some reference to set this up or test this.
Can anyone share me some sites as reference?
As far as I know there is no feature in Asterisk, but I might be wrong.
However, you can use ethereal to tap SIP connections. You simply sniff
the SIP connection and after it's done you can decode it and ethereal
will output a .au file which contains both sides of the conversation.
Also I heared that the Windows tool Cain Able is able to play back
SIP converstaions in real time, but I haven't tested that myself.
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[Asterisk-Users] Interesting article on new SIP phones
Hi all, Just a bit of news I picked up today http://www.theregister.co.uk/2005/06/02/computex_skype_handsets/ even though the s-word is mentioned, the handsets are also geared towards SIP. Cheers, Mike ___ 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] IAX blind transfers
[EMAIL PROTECTED] wrote: On Thu, 14 Apr 2005, Paul Seymour wrote: Just a quick question to ask if blind transfers (via #) are possible? I have an IAX2 connection to my VOIP provider. In my dial plan I sometimes forward an incoming call back out on IAX, but when this happens I seem to lose the ability to transfer the call. If the incoming call uses SIP, or the destination uses SIP, transfer works. I've noticed that in the Asterisk CLI I get message saying Attempting Native Bridge, but nothing more to indicate whether this failed or succeeded. I have tried notransfers=yes and notransfers=no in my iax.conf, but this doesn't seem to make a difference, as best I can tell asterisk is staying in the call though, so I'm guessing blind transfers aren't possible? If the Asterisk box IS doing IAX native bridging, then the # won't be seen on that box. Native bridging is a different thing than transfering. When an Asterisk box transfers the call it actual gets the two remote IAX peers to rather talk directly and sees nothing more of the call (in an IAX trace you'll see TXREQ (transfer request) frames and similar). notransfer is about enabling and disabling this feature. Native bridging is just where chan_iax2.c uses an optimised quick copy function to pass frames between the input and output iax connections. Native bridging will be done whenever two IAX channels are bridged together and both use the same codec etc. There's no config file option to enable or disable it. If you don't want native bridging, you need to disable it in chan_iax2.c by undefining BRIDGE_OPTIMIZATION. If you do that, then your box will probably hear and act on the # transfer request. Regards, Steve Hi Steve, Thanks for that explanation, it's very useful, I'm also having transfer problems on IAX2 bridges. In my case, I have a remote * with a couple of PSTN lines that are bridged over IAX2 to a local * which in turn handles a bunch of SIP phones. When a PSTN call is bridged, the SIP phones cannot transfer the call between them, seeing the behaviour explained in the first post. The IAX2 bridge is configured to use gsm, and no trunking. Could you confirm if this native bridge disabling could cure this problem too? Best regards, Mike ___ 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] Cisco 7960 SIP Firmware
[EMAIL PROTECTED] wrote: Cisco has recently changed the licensing distribution model for all of their phones. They are no longer currently selling the Spare version of the Cisco phones. I was told by Ingram Spain that they could only sell me the 'spare' version if I also purchased a CallManager license with it, which IMHO beats the purpose of it being called 'spare'. So, apparently, each phone is tied to it's license so-to-speak and the concept of 'spare' becomes rather vague. The new licensing program, as it was explained to me, will force distribution buyers who purchase any Cisco phones to also purchase a $150 SIP/MGCP license, this adds $150 to the list price of any model you purchase. If this is so, I expect to see Cisco phone sales decline. I was told by Cisco Spain that I had to supply the details of *my* end client to them, for quality assurance purposes, so that they can call the client and tell them how good a dealer I am (literally!). I imagine if I were to become a bad dealer, they could also phone all my client portfolio and direct them to an alternative good dealer. I ended up purchasing the phones from a distributor who didn't ask me any questions. In any case, it may well be the last Cisco phones I purchase. They are supposed to be releasing a new SP service provider edition of each phone model, which also will require the $150 SIP/MGCP license. I bet they wish we all pulled our trousers further up so they could tighten the belt and squeeze our necks a bit more. SNIP Perhaps there is a Cisco telephony authorized firm on this list who can shed some light on that seemingly illogical requirement. Er...Cisco's logic IMHO is inverted - I was also told by Cisco that they are now targeting small and medium-size bussiness, I presume because their growth potential in large companies is getting close to zero. I don't see how this policy, which seems clearly aimed at making you purchase their very expensive PBX solutions and their now more expensive phones in favour of cheaper PBX that can also work with their phones, ties up with the statements I got from them. Eventually, they are going to be fighting decent taiwanese imports with very cheap PBX systems, and I don't think many small or medium companies will have the slightest doubts on what is more cost effective. Regards, thanks for the information, Mike ___ 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] VoipJet Terms of Service
Jean-Michel Hiver wrote: But then again come to them with a few million monthly minutes under your belt and I'm sure they'll change the TOS for you... Maybe not, as the ToS also state: The customer agrees to purchase VoipJet termination in small amounts What does this mean? We have to start with 5-minute calls max, then slowly increase absolute timeout? How small is small? How about this one: VOIPJET DOES NOT SUGGEST, AND VEHEMENTLY DENIES, ANY CLAIM THAT ITS VOIP SERVICES HAVE A LEVEL OF QUALITY OR RELIABILITY ANYWHERE NEAR THAT OF THE REGULAR PHONE SYSTEM BWAHAHAHAHAHAHA this is like saying our system sucks and we know it. How can they seriously expect anyone that reads this ToS to want to sign up with them? It would have been simpler to simply state the usual we cannot guarantee 100% reliability or availability of our service, which depends on third parties over which we have no control or something along these lines. Nice laugh, best regards, Mike ___ 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] Calls from IAX2 trunk start again when hung
Hi all, I'm having a weird problem. The setup is Asterisk A with a TDM400P/4xFXO, DSL line, and Asterisk B with TDM400P/4xFXO on another DSL line. Both boxes are behind their own NAT. Asterisk B forwards calls from it's four PSTN ports to Asterisk A over an IAX2 trunk, which works fine using the GSM codec. Asterisk A dials the SIP phones on it's local segment. The problem is that when the inbound PSTN call ends, the hangups are detected, but for some reason, Asterisk B starts a new call all over again, Asteriks A receives it, the SIP phones ring, but when one of them picks up there is a dialtone, busy tone, or silence. Is there anything I may be missing here? I can post .conf files, but I don't think it has anything to do with those. Calls on the local PSTN ports of Asterisk A work fine. This setup is in Spain, FYI. Regards, Mike ___ 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] Calls from IAX2 trunk start again when hung
Rich Adamson wrote: I'm having a weird problem. The setup is Asterisk A with a TDM400P/4xFXO, DSL line, and Asterisk B with TDM400P/4xFXO on another DSL line. Both boxes are behind their own NAT. Asterisk B forwards calls from it's four PSTN ports to Asterisk A over an IAX2 trunk, which works fine using the GSM codec. Asterisk A dials the SIP phones on it's local segment. The problem is that when the inbound PSTN call ends, the hangups are detected, but for some reason, Asterisk B starts a new call all over again, Asteriks A receives it, the SIP phones ring, but when one of them picks up there is a dialtone, busy tone, or silence. Is there anything I may be missing here? I can post .conf files, but I don't think it has anything to do with those. Calls on the local PSTN ports of Asterisk A work fine. This setup is in Spain, FYI. Kind of sounds like an issue with detecting pstn line supervision events, but almost impossible to guess at root cause unless you provide something to look at. Might try some of the cli debug commands; 'zap debug', 'iax2 debug', etc. Look those over very closely and you're likely to spot the problem. If not, post the results. Include * version data as well. Hi Richard, Thanks for the pointers, I will try those debugs and will post the results. Best regards, Mike ___ 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
