similar to: SIP Problem (previous post) .. information might be relevant

OK, here goes my next problem. I have puchased a DID which I can connect to via SIP I have been given the following details: Username: uka1xxxxxx Password: 1000xxxxxx Server: brxxxx.net:5160 My equipment is Asterisk CVS HEAD on Red Hat EL 3.0 (NO NAT) The other end is a Cisco AS5300 (NO NAT) I can register with the Cisco with no problem. When I dial the DID it sends the call to my asterisk

i'm getting Asterisk to dial an h323 call termination service .. right now getting this message: -- Executing Wait("Zap/1-1", "1") in new stack -- Accepting call from '21382890' to 's' on channel 1, span 1 -- Executing Dial("Zap/1-1", "OH323/h323:723@216.52.153.206") in new stack 5:59.330 H323 Cleaner H323

Here's my sip debug output. anybody knows why Cisco sent * is CANCEL msg? Can someone tell me what ATA version are they using? Maybe this is also another issue.. I am using v2.16. This is using G711ulaw. Sip read: > SIP/2.0 100 Trying Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK13fbce3e From: "Jess" <sip:6882332@mydomain.com>;tag=as6818ebfb To:

Hello All, I have a media problem while using sip communicator user agent with asterisk behind NAT.I had enabled the debug mode in asterisk and capture the results.I have attached the results with this mail.Can any one help me to fix the problem? Thanks in advance, Partha __________________________________ Do you Yahoo!? Yahoo! Mail is new and improved - Check it out!

I have asterisk with following users; a) zaphfc ISDN card with two channels b) two mediatrix FXS gateways with four channels each c) 1x CISCO 7905G d) two notebooks with MS Messenger 4.7 Now, it seems that any combination works correctly in all combinations except when I call from MS messenger and then call is dropped always in 25th second of the call. Any ideas what I did wrong? here is my

Hello, I've been racking my brain over this for much of the day so I thought the list would probably be more helpful. A few days ago I upgraded from Asterisk 1.2 to Asterisk 1.4.5. Everything appeared to be working properly. However, on the first business day, we realized that when transferring calls (not using call parking, using the built in transfer buttons on a Cisco 7960) would not

Can someone help me what went wrong with this call? This call was initiated from dev/ttyI0 device on my asterisk server to mediatrix unit. Mediatrix unit user received the call and call started. I can hear them OK but they can not hear me correctly (cut-off sound, noise). Call was finally hunged up. Can anyone point out if there was something wrong? -*CLI> sip debug SIP Debugging Enabled

Hello. I have a strange problem. Its not possible to pickup a call that was placed with a Siemens SL75 Wlan. When this phone calls an internal number and i try to pickup (*8) the call from my phone i get nothing. It seems i have the call for one second or so but after that the call is being cancelled. No problems with other phones (polycom, grandstream). Attached the complete sip debug log

Hi everyone, Sorry if I'm repeating the e-mail, but I'm having problems with the list. I'm currently trying to enable call transfer to different domains in asterisk box (Asterisk 1.2.13 running on Debian etch). I have a configuration that requires me to transfer call to separate domains like ext at 10.10.10.10:5050. My calls come from a R2 channels in a board installed in the machine.

Hello everyone. I can place outgoing calls no problem with my IP500 (using teliax as our provider). Thing is, when a call comes in, 90% of the time when I pick up the handset it drops the call immediately. I turned on SIP debug, and have listed my extension config from sip.conf. Any help is greatly appreciated.... sooo close.... TIA! -Ron [3004] type=friend username=3004 password=XXX

Retrying, falling of the list some how :-( -------- Original Message -------- Subject: feeling n00b again Date: 2018-08-20 09:51 From: asterisk at a-domani.nl To: asterisk-users at lists.digium.com Hi all, Long time ago, I followed a Asterisk training, and both at work and at home, was able to deploy Asterisk, make all sorts of internal call (hard/soft voip-phones, incoming/outgoing,

Hello. I'm trying to use Asterisk in combination with SER, to make the routing proccess to my PSTN-Gateways. I made a simple test defining some extension in my extension.conf, when i made a call my SER (SIP) Server forward the call to Asterisk, this proccess is ok, but when the call is answered i see an INVITE going out from Asterisk to my SER Server, this invite is then passed to my

Hi all, I am using asterisk1.2.14,realtime and I find there is a strange case in the receiver's display. I have a dial plan to route a call to the destination. I haven't set the callerid(num) for the caller. In the receive ends, it's display shows "asterisk" when I make a call to the receiver. I wonder why "asterisk" shows in the display as I haven't set

Hi. We have an odd problem with incoming SIP calls. I have attached a SIP debug log, with some asterisk verbosity as well, demonstrating the problem, below. Is this a known bug? Vital stats: - Asterisk 1.2.3 - Sipura SPA-841, SPA-941 phones - Fedora core 3 The problem manifests itself with these symptoms: - an internal SIP extension receives a call from our PRI - the SIP phone answers the

Hullo :) I'm using Debian's Asterisk 1.0.7 bristuffed (though I'm only using CAPI for ISDN, and not HFC-S cards) and trying to transfer an incoming SIP call from sipgate.co.uk to any other extension. My phones are AT-320s (PA168S 1.43 firmware) whose documentation says to blind transfer, simply dial the number you want to transfer to, and press 'FWD'... This is what

I have the following in extentions.conf: exten => 2111,1,Dial(SIP/2111 at gw1.langley) exten => 2111,2,Voicemail(u2111) exten => 2111,3,Hangup exten => 2111,100,Voicemail(b2111) exten => 2111,101,Hangup I have the following in sip.conf: ; Cisco 1750 [gw1.langley] type=friend host=172.16.17.1 context=default canreinvite=no Like the ATA, lots of stuff doesn't work on the 1750

Here is the CS1000 log. Again, the CS1000 using SIP accepts incoming calls just fine. However, using outgoing call files the CS1000 is hanging up after I answer the call. I dont know why? Thanks, for any assistance. Jerry my sip.conf entry is: [Nortel] type=friend dtmfmode=rfc2833 username=XXXXXXXXX disallow=all allow=ulaw allow=alaw

Having an issue with receiving faxes, but when I pass through the fax. Currently, I receive the fax with Digium's Fax for Asterisk, store it and the initiate an outbound call to our fax server. (XMedius Fax). This works, but we would prefer to have Asterisk simply route the call directly to the fax server and take the store and forward out of the equation. When I do that, however, the

Hello, I have found asterisk is dialing a random number when it recieves a call, would anyone know why? The first thing I noticed found peer 4563 (this is a n Xlite Client) Many thanks, Spencer SIP Debugging Enabled spitfire*CLI> <-- SIP read from 82.70.154.145:5060: INVITE sip:448715046363@iptel.tgfslp.dalmany.co.uk SIP/2.0 Max-Forwards: 10 Record-Route:

REPOSTED with MORE Info and Modified Subject Line: -------------------------------------------------------- I am using one of the Minute Provider to dial out USA numbers. Now in one of my process, we need to Dial IVR and the enter DTMF digit and then it connects to the automated IVR. When I dial out the IVR directly using Xlite and VOIP Mins provider , it works perfectly. but when In try from