Re: [asterisk-users] Working softphone for poket PC

Hi try put Speaq speaQ is a VoIP softphone which runs on either Windows Mobile 5.0 or Sharp Zaurus Linux. It can be used to make and record Internet phone calls using any SIP compliant Internet Phone Server. The free Beta Trial Version which can be downloaded from this page, lets you record

Re: [asterisk-users] Working softphone for poket PC

Oops here is the link http://qtechinc.com/speaq_download.htm --Giridhar Bandi On 5/23/07, ram [EMAIL PROTECTED] wrote: On 5/23/07, Philipp von Klitzing [EMAIL PROTECTED] wrote: Hi! Googling arround I found a number of pocket pc softphones. Of those I was only able to install

Re: [Asterisk-Users] IVR sounds not on certain inbound route

I doubt this would be a codec issue . can you tell us what is the codec used on the sip and iax --Giridhar BandiOn 5/29/06, MC [EMAIL PROTECTED] wrote:Got 1 issue I can't seem to knock out of this particular box. The IVR works fine on the zap channels and the incoming SIP routes. Butcoming in via

Re: [Asterisk-Users] IVR sounds not on certain inbound route

please check if you are able to hear the sounds using alaw on IAX i had some problem listening to the sounds using G729 on sip client --Giridhar BandiOn 5/29/06, MC [EMAIL PROTECTED] wrote:Coming in on the IAX route is G729. On the SIP lines it is alaw.Giridhar Reddy Bandi wrote: I doubt this

Re: [Asterisk-Users] features.conf *1 Call Recording

did you include automon = *1 in your features.conf ?? it should be somthing like this [featuremap]automon = *1 --Giridhar Bandi On 5/12/06, Dave Morrow [EMAIL PROTECTED] wrote: Thanks for the response.How would I change the DTMF transfer mode?David MorrowTechnical Systems LeadAutodata Solutions

Re: [Asterisk-Users] features.conf *1 Call Recording

hi Dave i get the following log on *CLI -- Attempting native bridge of SIP/200-39f4 and SIP/204-2ce4 -- Playing 'beep' (language 'en') -- User hit '*1' to record call. filename: wav|auto-1147452537-200-204|m -- Playing 'beep' (language 'en') -- User hit '*1' to stop recording call. --

Re: [Asterisk-Users] IVR: playing multiple streams simultaneously?

hi i don't know if we can do that . but i guess we can use audacity .. to mix both the files and get what you want.-Giridhar BandiOn 4/18/06, Herchi Silviu [EMAIL PROTECTED] wrote: Hi all, I'm setting up an IVR using Asterisk. Is there a way to have two streams played to the caller

Re: [Asterisk-Users] Voicemail problem

Hi Daniel can you give us more information so that it would be easy to debug.like voice mail configuration etc Thanks,GIridhar Bandi.On 4/18/06, Daniel Korndorfer [EMAIL PROTECTED] wrote: Hi,when I call the voicemail app, it starts and die suddenly. Has anyonealready had this

Re: [Asterisk-Users] Asterisk service crashes

Hi Billy Try using safe_asterisk and see . safe_asterisk be useful if you fear asterisk may crash.--Giridhar BandiOn 4/19/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: List, The past few days the asterisk service on my server has crashed several times. I have had it running for

[Asterisk-Users] polycom unable to start recoding

Hi i have a Polycom Soundstation premier Basic Conference Telephone which is connected to Linksys pap2 boxi am unable to start recording using *1 . but with a normal analog phone connected to linksys pap2 i am able to start and stop recording .. i tried changing the DTMF setting but no use . can

[Asterisk-Users] Asterisk IVR / Scalability

Hi i am looking for a good ivr system for my company. these are my question are there any good ivr's that can be easily integrated with asterisk ? and are there any large scale deployment of asterisk to date ? thanks Giridhar Bandi ___

Re: [Asterisk-Users] Asterisk behind NAT

IT works fine behind firewall .enable NAT in sip.conf and it works fine.Giridhar BandiOn 4/6/06, Joao Pereira [EMAIL PROTECTED] wrote:Hello to allCan we put Asterisk in a company that has an ADSL connection with just one public IP address? Because with just one public IP, Asterisk musthave a

Re: [Asterisk-Users] Asterisk behind NAT

so that means that a sip client can access asterisk server which is behind NAT ( assuming that SIP and RTP ports are properly farwarded ) even is nat=no in sip.conf thanks,Giridhar Bandi. ___ --Bandwidth and Colocation provided by Easynews.com --

[Asterisk-Users] DID registration status

HII have two sip accounts from two different ITSP's both configured on asterisk server. how can i know if these accounts have been successfully registered ?i generally look at the /var/log/asterisk/full suggest me if there are better way of doing this thanksGiridhar Bandi

[Asterisk-Users] How is Teliax ?

Hi I am looking at purchasing some DID lines from Teliax to install it on my asterisk. i would like to know some feed back on Teliax before i purchase. suggest me if there are better sevice providers. thanks Giridhar Bandi ___ --Bandwidth and

Re: [Asterisk-Users] How is Teliax ?

thank you all for the feed back. --Giridhar Bandi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users