similar to: SIP - Problem with audio clipping

I'm experiencing a very strange problem with SIP calls with a CLEC (CBeyond). The downstream audio with the telephone on mute is excellent. However, when there is upstream audio (even breathing) from the mic, the downstream audio is clipped and sometimes dropped. The strange thing is, if I Monitor the call, the downstream audio in the wav file is perfect, even though there was clipping

I'm experiencing a very strange problem with SIP calls with a CLEC (CBeyond). The downstream audio with the telephone on mute is excellent. However, when there is upstream audio (i.e., breathing) from the mic, the downstream audio is clipped and sometimes dropped. The strange thing is, if I Monitor the call, the downstream audio in the wav file is perfect, even though there was clipping

I'm having difficulty with an Asterisk system. The external party has very good call quality, but the internal party hears clipping and drop outs. The WAN comes in from the Cisco IAD and into a LAN switch (DLink DGS-1005D w/ 802.1p) where the two public IPs are switched to different devices. One device is a FireBox device controlling a separate LAN with VPNs. The other device is eth0

I ripped a rock-and-roll CD for a client's moh. But it's too loud. Is there a simple way to reduce the gain without having to remix the tracks? Thanks -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 mike@TelecomMatters.net www.TelecomMatters.net

I'm seeing quite a few "Not Found" pages when I google lists.digium.com. Is anyone else getting this? -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 mike@TelecomMatters.net www.TelecomMatters.net

When I look at my CDR data for calls to NuFone, the billsec for each call is 14 seconds or less. When I look at my NuFone account, the billsec has normal call lengths. So it seems that the billing on the Asterisk system terminates after about 14 seconds. The calls come in on an IAX connection and go out to NuFone on IAX. Are these calls bridging away from the Asterisk server? How can I

The attendant attempts an attended call transfer (all phones are IP501). The attendant pushes "hold", "transfer", dials the extension and announces the call. When the attendant pushes "transfer" the second time, the original call is lost. The reason this is a big problem is that the PRI channel for the call remains busy. Subsequent inbound calls on that

Is there a programmatic to to trim the silence from the beginning and end of a recording? From a .wav file? From a .ulaw file? Thanks, -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 mike@TelecomMatters.net www.TelecomMatters.net

I would like to capture the lat/lon coordinates from a GPS-enabled cell phone or PDA. Is this possible? Must I subscribe to this information from the cellphone network provider, or can I capture it without charge? What devices will broadcast the coordinates? Is there a device that will broadcast its position inband that can be captured by Asterisk? Can an SMS message include coordinates?

I need to test QoS on an IAX link between a server in Colorado and a server in Europe. I know I could install a Milliwatt extension on the European server and just listen, but is there a more scientific method to collect QoS metrics? Thanks P.S. I'm getting a lot of "Page Not Found" on lists.digium.com. Are the older posts being purged? -- Michael Welter Telecom Matters

My CLEC just called and asked if we will support the "MI2" protocol on our proposed T1 circuit. I think this is for CallerID name. Will the T100P support this? Thanks, Mike

Did Linksys really use Asterisk for the SPA9000 software? -- Andres Technical Support http://www.telesip.net

When I bought two Polycom 501 SIP phones, I naively thought they were Power-over-Ethernet (IEEE 802.3af) because they were "powered over ethernet." Silly me. Polycom must have some odd voltage or funny way of injecting the power, because the POE switch I bought for them (Netgear F@510P) won't power them, though if I use the Polycom-supplied AC adapter and ethernet power

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 -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is there a Linux tool that will splice several gsm sound clips together into one clip? In my agi script, I would like to use 'get_data' with one clip instead of multiple 'stream_file' so the user doesn't have to listen to the entire spiel before responding. Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 mike@introspect.com

Greetings folks, I'm currently dealing with a company to let me set Caller-ID-Name on outbound calls. So far pretty happy with their services. The basic service works like this: * CLEC sets Point Code to point to this company * CLEC has to sign LOA saying they give me permission to set the Caller-ID-Name through this company. * I go into web interface and set name. However, the CLEC is

I have the following configuration: CLEC -> T-1 -> Asterisk -> Adtran Channel Bank -> (analog) -> Nortel Don't complain that it's ugly. I've already done plenty of that. The CLEC manages their Adtran remotely and needs to be able to continue to do so. I assume they use FDL to do the management. We are using the Zaptel DACS/DACSRBS to cross connect some of the

Hi, My PBX/key system failed when building power was switched off and then back on. The service rep says the PBX unit is so old that it is not repairable. The unit has 8 incoming POTS lines and 12 multiline sets. There are voice mailboxes on each line. Switching to Qwest's Centrex system would cost about $3800/yr. My 8 business lines now cost $3320/yr, so the Centrex increment would

Does anyone know of a 24 port ATA device that could be installed in a phone closet? Like a channel bank, but, instead of multiplexing onto a T-1 circuit, it would convert to SIP/RTP on a LAN connection. Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 mike@introspect.com www.introspect.com

Hi all, I am trying to setup a ZAP interface to do MF signaling for a handoff to a 911 tandem. The signaling I need to perform on the T1 is this: 9-1-1 Tandem: Wink CLEC end office: KP (Keypulse) NPA ST (Start) 9-1-1 Tandem: Wink CLEC end office: KP I (Info Digit) NXX XXXX ST As I'm not as familiar with the Zaptel configurabliity, I'm not really sure how to do this. Do I dial twice