similar to: Asterisk and Cisco 7960 problems persist (for me, anyway)

After downloading the latest CVS head and testing it with the Cisco 7960 (SIP v7.2), it appears that it's still necessary to patch rtp.c to avoid audio dropouts. I'm quite sure my gateway provider is running an older version of Asterisk, and I suppose that this may be the root cause. But I mention the issue here because it seems like it would be a mistake to ship Asterisk 1.0 if it

Hi, I'm experiencing some voice delay (2-3 sec) after outgoing call is setup. It means during the first 2-3 secs, audio is very choppy or nothing. So usually I can't hear the 'Hello". I use IAX2 for my ougoing calls with Grandstream phone as a client. Any hints to prevent this? Thanks, David

Hello, I'm considering using Asterisk with some type of Cisco phone, and currently considering either the 7940 or 7960 because of its more-complete functionality (compared to the 7905). I'm currently wondering: Do all the expected functions (transfer, conference, voice mail, message waiting indicator, etc.) work normally with Asterisk over SIP? What caveats are known about using

What you and so may others on this lise seem to forget is that Cisco is a company offering bsuiness products for businesses. Businesses typically pay by check and wire transfer, especially for items such as this. If you want home-user pay-by-credit-card service, buy products from Belkin's home line and similar. Oh...what's that? None of these cheesy Stocked-at-Costco hardware

All, The users in our organization are well, quite frankly, sick of phone service that is being provided. The choppy phone calls, and drop outs are detrimental to our sales force. I've tried about everything I can think of. Moved the asterisk server from VM machine to dedicated machine More than enough bandwidth Setting 802.1p = 7 Set Dedicated voice traffic 35% of bandwidth. Not sure

For those that might be using Cisco 7940/7960 sip phones and placing calls across an iax2 link, we think the voice quality problem has been identified and corrected. The dev cvs should be updated as of about 3:30pm CDT today (April 14). History: Calls originating from a Cisco 79x0 sip phone and sent via iax2 link to some distant * machine resulted in very poor quality audio, and in some cases,

I am getting ready to install Asterisk and I was looking into the Polycom IP600 phones. I spoke with Polycom sales to verify the multiple line appearance and they said it would work. More specifically, if lines 1-3 all contain the same SIP registration info, the Polycom will only send out 1 SIP registration to the server and then handle the calls ringing on multiple lines. I was wondering if

Okay everyone - I'm moving away from using sipura 841 phones. I'm starting to test with Polycom IP 501 phones. We plan to upgrade our server to a dual t1 but, for now we have a digium tdm400P with 4 analog lines coming into it. So my question is will upgrading the IP phones with my existing digium tdm400 card be enough to satisfy my users ? or is it really a combo deal needing to

I upgraded to the latest HEAD version of asterisk, and all IAX calls started sounding choppy. It was suggested on the IRC channel that I go back to asterisk -stable to determine if that fixes it. Is downgrading as simple as upgrading? Because now, -stable builds fine, but I get an error on the asterisk console when starting, something about "ast_get_txt" not found. Recompiling and

Should FAX transmission generally work through Asterisk and a TDM400P connected through a PSTN gateway? At first blush I'd think that if they're all g.711uLaw encoded that it would work. But experience shows otherwise. Is there a better way to do FAX? -brian

Let's say that I have a call coming in to Asterisk through a TDM400P and going out through SIP to someone on the Internet. Is there any configuration option that would allow me to do silence suppression on the RTP stream generated by Asterisk on behalf of the TDM400P connected user? SIP phones allow me to do this easily, but I'd like to be able to conserve upstream bandwidth on calls that

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031012/59c113df/attachment.htm -------------- next part -------------- Hello, We are planning to buy following Hardware for Asterisk TestBed. Please let me know if this seems fine to you. 1. IP Phones ( 5 in number) CISCO 7940/7960, SNOM 200, Pingtel xpressa 2. Wildcard T100P interface card,

Hello List, I'm trying to prepare some lecture notes on non parametric methods, and I can't manually reproduce the results of the wilcox.test function for ordinal data. The data I'm using are from David Howell's website, available here http://www.uvm.edu/~dhowell/StatPages/More_Stuff/OrdinalChisq/OrdinalChiSq.html If I run the wilcox.test function on the data I get a p-value of

Hello. I have another issue. When I call in, everything is processed correctly, including voicemail, but I don't hear any ringing/ringback. exten => s,1,Zapateller(answer|nocallerid) exten => s,2,NoOp exten => s,3,Playback(pls-wait-connect-call) exten => s,4,Dial(${PHONE1}&${PHONE2}&${PHONE3}&${PHONE4},15,Ttm) exten => s,5,Answer exten => s,6,Wait(1) exten

Hi, After following the suggestions on the mailing lists and the wiki I'm still experiencing choppy moh. The song plays but with frequent noise parts. - I'm using asterisk 1.2.4 on our production server and 1.2.7 on the test server. - native moh with .gsm and .pcm formats (according to http://astrecipes.net/?n=152) - compiled ztdummy as a timing source any pointers on how to dig deeper

Hi, Is there is some form of VAD on * for SIP channels, cause I have a problem with MOH. I made an extension which simply plays MOH, when I dial that extension with my ATA188 MOH sounds choppy if I talk on the phone the MOH keeps playing. I saw the sip channel (show channel SIP/*) and I see no packets going in/out when I talk then packets shows going in/out. I don?t have this kind of problem

Hello, We are at wit's end on this. One (and only one) of our five asterisk installation is giving us real headaches. Buzzing and/or choppy sound interfere with conversations. I recorded some conversations with monitor() and no problem whatsoever appear in the recording, while the local user was hearing the buzz and half my words. This is a 1.2.16 installation with mISDN but mostly using

Hi all, We have recently implemented an Asterisk system using Trixbox (asterisk v1.4.4 at the moment, yet to move to 1.4.9) but are getting pressure to switch back to our old key system unless we fix two major issues. So please help me avoid switching back! An overview: We have about 12 Linksys SPA941 SIP phones connected on a private switched network to our asterisk box which is a

Hi, I am setting up a dispatch center where will have 4 call takers, all with Polycom IP 600 Sip phones. Each phone will be setup with 6 extensions each. When a new call comes in, the first extension on all the phones will ring. This works fine, the problem is when one of the dispatchers is already using her first extension and another call comes in. What happens now is that the remaining 3

Has any had any experiences with Watchguard Firebox 1000 and Asterisk. I have asterisk on public side and phones on the private side. I am able to get the phones to register and make outbound calls but the inbound calls are intermittent. I have NAT enable in asterisk and on the Cisco 7960. Any insight would be appreciated. Thanks