similar to: Re: poor state of IAX2 code? (was: why a per fectlyfine iax2 host becomes UNREACHABLE?)

If you are a Canadian VoIP provider or CLEC, help make a difference by joining the CAVP: www.cavp.ca >OMFG, >I thought April Fools day was over. This is hard to believe. If true it >tells me that Bell has not changed at all. They are still trying to >manipulate and take advantage of the parts of the market they have absolute >control over. The CRTC was right to continue to

I've got this low-ping 100%-up dsl connection between two asterisk 1.2.7.1 servers. And oftentimes one of them would declare its opposite UNREACHABLE. Why can this happen? The host stanzas in iax.conf have raw IP's, so no DNS monkey business here.. An inquiring mind wants to know.

Dear Royce; Did ur problem resolved? Because now I am facing same problem. It look like that it happens with IAX trunk only, but does not happen with IAX endpoints that registering (as trunk does not register, it sends the call directly). My initial analysis that one of the following can help to let the trunks talk: if there is an IAX endpoints registering to the machines, then trunk become

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

Using a SIP connection with a CLEC, the downstream (received) audio is perfect when the mute button is activated on the phone. However, when there is upstream audio (i.e., talking or even breathing into the microphone), the downstream audio is cut off. It's kinda like having a half-duplex audio connection. When I divert outgoing calls to another provider, these calls are fine.

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

My client would like to add new functionality, such as call queuing, to their PBX. They presently have a Telrad PBX; it has a T1/PRI circuit from the CLEC and a few POTS lines. They use DID for some incoming calls and an attendant for calls on the main number. The client's commitment to customer service will not allow us to outright replace their PBX with a new system--we could never

I have a network of offices using Asterisk that are connected via IAX2 trunks. The trunks work great for a day or two then for no reason at all one end of the trunk will become UNREACHABLE while the other end is still connected. The only way to fix the problem is to shutdown Asterisk completly then start it backup again. The end that dies is not always the same, some times it is server A and some

Thanks Brian, and thanks again for the included definitions <grin> - that helped too. Your comments are really helping clear many questions. I suppose our intensions are to become an IXC. So if my local carrier is sporting old technology, they'll provide TDM services. So if I understood you correctly, the "in-band signaling" is typically SS7, and the alternative is typically

Hello, I recently upgraded from asterisk 1.4 to 1.6. I am using the same SIP settings in sip.conf in this version also. I am facing a problem when a SIP client makes a call. When a SIP client registers to asterisk its status shows 'OK' and it is able to receive incoming calls. But as soon as this client make a call, its status becomes 'UNREACHABLE' and it cannot receive any

Hello, we have the problem: We run 7 blades with a xen 3.0.1tha3 configuration (with the patches for debian from debianbase.de), which a is xenisied 2.6.12.6-kernel. Everything runs fine until we start producing heavy load on the network (e.g. multiple clients access our maxdb database), the dom0 and the domU network becomes unreachable (i.e. only every 100th-200th ping gets through).

I have a situation where we have an asterisk box that is extending several Mitel PBX extensions to some cordless SIP phones (Cisco WIP310). Everything works great, except when the cordless phone walks out of range of one access point and into range of another (cisco 1100 series APs). I have another post to the list asking about how to speed up the handoff, and keep the call active while

Hi all, I'm trying to get a handoff between me and a carrier going using Asterisk. I need to handoff using CAMA signaling. On a Cisco, you can see the configuration types that I'm referring to on this site as an example: http://www.cisco.com/en/US/products/hw/routers/ps221/prod_configuration_guide09186a008019b16e.html#35393 The example of the interaction is the following: CLEC end

I've watched the dialogue about how asterisk has to manipulate the packets from an IAX2 connection to a SIP client. That said, I'm wondering if a previous problem I've been trying to diagnose could be related to that process. In short, here's how I describe it: Outbound: SIP 7960 > Asterisk > IAX2 Audio is perfect both directions. Inbound: IAX2 > Asterisk > SIP

Hi, Is my CLEC feeding me a line, or is this really how it has to be? When I dial a number on their system, if it's busy I get a busy back right away. If I dial a number on say Verizon's system, the CLEC sends me a "call preceding event", and then I get a BUSY back from Verizon. Should the CLEC be able to only send me ahead if Verizon says "go head" and give me a

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

> Is anybody on this list actually using iax2 for > anything mission-critical? Yes. 2K inbound / outbound calls a day to 30 remote locations, aggregated to 2 PRI's tied together with IAX2. All with IP address specified rather than hostname. All with Asterisk 1.0.9. All with 99.9% completion rate, and it would be 99.999% if we weren't using consumer grade DOCSIS cable modems in the

I've turned on debug in a (IAXComm based) soft phone. I see the phone sending pings to *. I see * getting the pings. For some reason, with iax2 show debug, I never see any response on the console from *. However, the phone shows a response with INVAL. Seems like an odd response to a ping request. I believed it should get an ACK. Is that wrong? If I should get an ACK, what could I have messed

I would like to ask, if someone observe also problem with peer qualify problems, my asterisk log is full with UNREACHABLE/REACHABLE messages, even when two asterisks are in LAN environment, please take a look into this debug, I can't find any problem with packet loss, all qualify requests are replied and acknowledged, I will submit bug report, if you will also not find any problems here...

I know this is not strictly an asterisk issue but it is related I guess. Just to let you know that after many calls to Mitel the consensus is that they will be releasing a new version of the 5220 that is dual boot (minet and SIP) next week or the week after. This firmware will only appear on NEW phones manufactured after the release date (no one could confirm but the 23rd of sept was