[Asterisk-Users] MixMonitor ControlPlayback of g729 files

Hi Should Mixmonitor ControlPlayback suppport file recordings in g279 format (I have enough licenses). call is alaw to alaw but would like to store the calls in g729 format instead of gsm. Thanks __ Do You Yahoo!? Tired of spam? Yahoo! Mail

[Asterisk-Users] SIP compact headers

Hi Anyone know if parsing of SIP compact headers is slower than full headers? like if there is an extra lookup step - mapping short to long? or should it be faster or about the same? Thanks __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the

[Asterisk-Users] Emergency calls - forcing through on channel

Hi Hypothetical but quite possible scenerio: Attempted emergency 911 call but all zap channels are already in use. Is there any way to hangup zap channels before dial(Zap/g1/911) or equivalent. AFAIK hangup doesn't except options so I CANNOT do something like hangup(Zap/g1) to clear zap

[Asterisk-Users] Emergency calls - forcing through on channel

Ideally: - define emergency number - if number is called in the dialplan and all available channels are busy within the given technology (like ZAP) - hangs up then dials out on the first channel that is NOT currently on an emergency call __

[Asterisk-Users] Emergency calls - forcing through on channel

Thanks Brent I must have missed it on the wiki! At first glance the last example is close to what i want even though it looks a bit complex. A native - buit in - emergency number feature for asterisk would have my vote. Your article is: http://www.voip-info.org/wiki-Asterisk+tips+911

[Asterisk-Users] agent channel busy - how to stop it?

when a call file is used to place a call FROM an agent the agent is flagged as busy/unavail even if the call is subsequently transfered. call file has...Channel: AGENT/blah... Any way to stop the agent channel being flagged as busy? Cheers __

[Asterisk-Users] re: call load balancing

just my .02 2 or 3 shabby DSL connections and use asterisk to monitor the quality of each connection then route calls according to the best option at any given time I really think this is going to be a false economy would compared to the cost of an SDSL line with an SLA. Also network issues

[Asterisk-Users] Calling Extension directly

or you could have a tel (or DDI) number for each internal extension giving each 1 a real tel number. __ Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard. http://promotions.yahoo.com/new_mail

[Asterisk-Users] audio fading in and out?

Hi I am trying to find a possible cause of 'audio fading in and out' which effects about 0.1% to 0.15% of calls placed to a voip provider for termination. (I am using SIP alaw throughout in this case) I don't believe this to be something that is network / jitter etc related (?) and looking

[Asterisk-Users] IDSN 30 PRI UK

Hi I am ordering a ISDN 30 line in from BT to use with digium hardware. Was wondering if there was anything specific I should ask for when getting the service in place. Thanks Start your day with Yahoo! - make it your home

[Asterisk-Users] chan_agent / manager API / SIP - possible bug?

Hi Was wondering if anyone could confirm this? agent is logged in using agent callback login to a sip extension when placing a call with the API action: originate and the channel: agent/blah the call only works one time then seems to tie up the sip channel. when placing a call using a call

[Asterisk-Users] CVS Build from 16-7-2005 Crash! bug or what? ; -D

Probably doesn't help diagnose the problem but there were also audio problems experienced with this cvs version even on LAN / sip2sip / no transcoding ERROR[1171] UTILS.C:509 TVFIX: WARNING NEGATIVE TIMESTAMP -194931. ... I will be looking into this issue later today.

[Asterisk-Users] CVS Build from 16-7-2005 Crash! bug or what? ;-D

i get lots of the below from friday 15.7.05 cvs as well ERROR[1171] UTILS.C:509 TVFIX: WARNING NEGATIVE TIMESTAMP -194931. ... Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs

[Asterisk-Users] Polycom Auto-Answer problems

you can also use answer as the ring type instead of ring-answer if you just want it to pick up. I would keep Ring_Ans the same throughout for simplicity exten = 301,1,SetVar(_ALERT_INFO=Ring_Ans) add an alert info type (say type 5) alertInfo voIpProt.SIP.alertinfo.5.value=Ring_Ans

[Asterisk-Users] auto dialing - call file - channel variable question

as it turns out if the agent (with callbacklogin) is logged into SIP/123 it will try to dial 123 in local context. only variable I can get so far that is passed on is callerid If i want auto answer I set caller id to include a specific string In local context I have a dialplan matching 123

[Asterisk-Users] auto dialing - call file - channel variable question

Hi When using a call file to place a call I can't seem to figure out how to get the variable alert_info passed to the actual channel (in my case a SIP phone) that an agent is logged in at. Please can someone give me a pointer in the right direction ;) Thanx! Probably best illustrated in an

[Asterisk-Users] isdn30 / pri lines in the UK

anybody recommend a supplier in the UK for a pri/isdn30 line (other than BT) thanx very much __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___

[Asterisk-Users] Polycom distributor in the UK ?

according to Polycom the IP301,IP501 are not going to be released in the UK (EMEA) until Q4 this year... try calling hardware.com if they have them available. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around

[Asterisk-Users] 12 seat call centre with Asterisk, VoIP only, UK - possible?

Talk to BT about getting an ISDN30 line put in... you'll get some sort of guaranteed quality and it'll be much better than a pure SIP solution. Talk to anyone APART from BT, their pricing is much more than OLOs etc. Who else can I order an ISDN30 line from in the UK?? I am looking for one at

[Asterisk-Users] AgentCallBacklogin (logout continued...)

Hi Finally decided on an ugly workaround for my logout with password problem - storing the agent passwords as variables. If anyone has any better ideas or suggestions please let me know... -- in this setup agent number is 2000, agents are 2XXX and SIP phones are 1XXX main bits of

[Asterisk-Users] AgentCallBacklogin (logout continued...)

Thanks for the info alan unfortunately I am trying to logout an agent that has a password. Example did give me ideas on how to do some other stuff though. I agree completely it is kind of silly to require a password to logout. Anyone know if: there is a way to execute something like the below

[Asterisk-Users] AgentCallBacklogin (logout continued...)

Anyone know if - it is possible to limit 1 agent per extension where the last agent to log in overrides any previous agents or - a Command/application to clear all agents logged in on extension Does this look like it would require a custom mod to do it? J

[Asterisk-Users] jitter buffer recomendations

Hi I seem to be getting about 250-500ms drop outs on receive audio for some calls that are routed over the internet. Probably no news there. My end point is a polycom 500 and was wondering if anyone could recommended jitter buffer settings. codec is alaw min buffer is at 40ms maximum buffer is

[Asterisk-Users] AES vs AEC

Hi Does anyone happen to know the difference between echo cancellation vs echo suppression - particulary in relation to polycom settings - the sites I have come across seem to use the terms interchangeably. I seem to be one of the unlucky few getting a very slight echo on lan calls. Thanks

[Asterisk-Users] SIP Jitter buffer

Hi I am using CVS latest Is it correct there is no jitter buffer for SIP (RTP) Are there any plans for this? prob a stupid question: Is it required / do the endpoints handle this - if the src and destination are both SIP and there is no transcoding but asterisk is still in the media path?

[Asterisk-Users] Maybe an echo cancellation problem?

Hi Was hoping someone could point me in the right direction. using asterisk cvs in various VOIP configurations On a call when the loudness of transmit receive then all receiving is null. In practical terms this causes background noise (from the other end)to stop when you are talking and

[Asterisk-Users] Maybe an echo cancellation problem?

Thanks for the pointers. I have confirmed that Voice detection is disabled and echo cancellation/suppression settings are disabled on the voip client (Polycom) The best way to describe it is half duplex audio when transmit is greater than receive. Anything else I could look into? I also