similar to: How to tell the diff. between a fax and an audio call on outbound calls

All; I'm having a problem sending an outbound fax using Asterisk-1.8.15-cert3 and the spandsp fax module using a SIP trunk. I'm seeing hundreds of these: ERROR[14423]: udptl.c:294 encode_open_type: UDPTL (SIP/runcentral_outbound-00000074): Buffer overflow detected (59 + 134 > 175) Has anyone ever seen this before? I have the following configuration. udptl.conf:

All, I was wondering if anyone has any experience with the Cisco ATA186 (SIP image) and outbound faxing with Asterisk. Inbound faxs from PSTN into * and on to the ATA work fine but outbound faxs receive congestion from *. I've got packet dumps from both sides and everything appears normal but after about 3 seconds the * servers sends the AS5300 a CANCEL and sends the ATA a '503 Service

Hello all. I have the following setup: Fax machine | Sipura SPA-3120 | SIP 100BaseT | Asterisk 1.4 | IAX2 100BaseT | Asterisk 1.6 | ISDN PRI TE210P | Traditional Telco The fax lands on the Internal Asterisk 1.4 box, the sip config for this extension looks like: [35081] type=friend secret=************ qualify=yes port=5060 nat=no host=dynamic dtmfmode=rfc2833

All; I have a few customers that do a lot of faxing, both inbound and outbound. Some use the Spandsp and some use the Digium FFA modules. What I would like to do is when an outbound fax fails, determine whether the remote end was a fax machine or a plain old phone. I would also like to determine that in the dial plan if I could. Any insight at all with this would be extremely helpful.

Hi I need a small PBX for use on the move. This means that outbound calls will need to be made over the cell phone network. Assuming a small hardware PBX with a spare mini-PCI slot or a USB slot then what hardware options do I have to get an outbound cellular channel? Options need to be rock solid, so no bluetooth to a cell phone kind of solutions need apply. Can any of the 3G usb

List members, Is it possible to record outbound analog calls using an X100P? I was asked if I knew how to record all calls for a shop with 4 analog phones transparently to the end users. I thought Asterisk was a good fit for this and I envisioned using either Digium TDM400Ps or Sangoma A200s with 4 FXO and 4 FXS modules. The FXO modules would be connected to the existing PBX and the FXS

Hello! I've run into a problem where a user is making an outbound call at the same time that an inbound call is being made on the same analog line. It appears that as the zap channel is opened for the outbound call, it is simply answering the inbound call. Obviously, both parties involved in the calling get a bit confused. Previously, it happened only on an occasional basis. However, as this

I am trying to make outbound calls from my Asterisk client through a remote Asterisk server with IAX. In iax.conf on both sides [dar] context=trusted secret=xxxxxx type=friend host=192.168.1.1 in extensions.conf at the client making the call Exten=_1NXXNXXXXXX,1,Dial(IAX2/dar:xxxxxx@192.168.1.1/) What goes in extensions.conf at the remote server? What is needed for the

So, I'm still having this problem with outbound calls not working when using a channel bank. I've purchased a Rhino FXO channel bank from VoIPSupply.com to make sure it wasn't an equipment problem. I am using a Digium TE411P card, and have simplified it down to just 1 port plugged into the channel bank, with just 1 analog line plugged in. If I place an inbound call on the line, it

The host I am working with has two accounts from the same DID provider. Incoming calls work correctly and dial the appropriate extensions. This also allows incoming calls to be billed appropriately to the individual DID accounts. Outgoing calls from either extension default to the first DID, i.e. calls from either extension have the same callerID. How can an extension specify separate outgoing

Hi all, a noob here, I am trying to get outbound calls through asterisk working with Broadvoice. I have consulted the following two online tutorials: http://www.broadvoice.com/support_install_asterisk.html http://www.voip-info.org/wiki/view/Asterisk+settings+Broadvoice in an effort to make outbound calls. My current settings are as follows: sip.conf register =>

Hello All, I would like to record inbound and outbound calls to and from one number. I tried to add lines to my extensions.conf: DAY=`date "+%m-%d-%y_%H:%m"` ;outbound exten => 5555551212,1,Record(${DAY}:gsm) exten => 5555551212,2,Dial(${TRUNKL3}/${EXTEN}) ;Inbound [line2] exten => 5555551212,1,Record(${DAY}:gsm) exten => 5555551212,2,Dial(SIP/101,20) exten =>

Hi All, Simple scenario: 7940 SIP><NAT Router><INTERNET><Asterisk SIP B2BUA w/Public IP Inbound/outbound calls work fine 2 way audio, features ok, no issues that I can tell so far. 7940 SIP Using Outbound SIP Proxy><NAT Router><INTERNET><Asterisk SIP w/Public IP Phone registers, call in/out SIP Signaling traversing the proxy ok no audio on phone, SDP

Hey everyone ! I have a problem in making outbound calls in PRI connection. I have E1 PRI airtel connection [ India ] [ asterisk-1.2.12.1 on CentOS 4.4 ] zaptel.conf ---------- [channels] language=en usecallerid = yes hidecallerid = no callwaiting=yes threewaycalling = yes usecallingpres=yes transfer = yes echocancel = yes echotraining = yes immediate = no ;group=0 ;context = from-pstn

OK, been messing with Asterisk for a long time and I have my opinion on where the issues lies but sometimes it's just nice to see what others think that can relate :-) Here goes.. Inbound calls flow like this:Tier 1 Provider (SIP) > Asterisk 1.8 > Name Brand PBX - Calls work fine Outbound calls flow like this:Name Brand PBX > Asterisk 1.8 > Tier 1 provider (SIP) - Calls work fine

2 Asterisk servers 1.2.12.1 Connected via IAX2, same switch, GigE, no packet loss, etc 1 with a Sangoma A101 for a PRI to the PSTN Ulaw QoS enabled NAT for the registered ATA boxes, no nat between the * servers Faxing inbound: Call from PRI hits the first Asterisk server Then talks to the 2nd via IAX2 NVFaxDetect receives the fax, converts to PDF and emails it out Works great!

Hi. recently I have been trying to setup a PRI on asterisk. Inbound calls are working just fine but I am not able to make outbound calls. Does anyone know what I need to change to make outbound calls work? Right now the PRI is instantly hanging up on the outbound calls. I have included full debug info as well as config files. /etc/zaptel.conf span=1,1,1,esf,b8zs bchan=1-23 dchan=24

I put a post on here about my issues with outbound calls not ringing but i haven't resolved it. so i am trying again. When i dial any outside number i dont get a ring tone at all. when the person picks up and starts to talk i can hear them fine. it sounds great. How do I start to troubleshot this? _________________________________________________________________ With Windows Live, you can

Is there a way to limit the number of simultaneous outbound SIP calls made by Asterisk? We've tried using the 'Asterisk sip call-limit' parameter but that doesn't seem to be working and one of our engineers says that parameter has been depreciated. Thanks, Deric.Page at nisc.coop -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm running a pretty successful Asterisk system and recently moved our PRI to a T100P board. The PRI was previously connected to a Cisco 2600 that was serving as a voice gateway. We are having a frequent problem with inbound and outbound calls being disconnected shortly after they are answered since moving the PRI directly to the Asterisk box. Most calls work fine, but approx 3 out 10 are