similar to: help on dial plan

voipstunt: "Forbidden - wrong password on authentication for INVITE to ...." I have paid, the password was not changed, ... I have no idea why. Is there anything what I can do to get this "failed" call over to another provider, so that the user can complete the call? (Dialstatus was an idea, but the line does not show up in CLI) [Apr 5 09:22:36] -- Executing

How to set-up LCR ? a. which companies can be used with LCR? b. how to set-up & maintain LCR? c. multiple connection to one gateway? Example: +886223456789 could be reachable via a. ENUM free b. Dundi free c. Voipstunt free d. Voipbuster free e. Nufone $ f. Voipstunt $ g. others with 4 concurrent connections $$ h. others with 3 concurrent connections $$ I am looking

Hi, all I am reallty pissed with their service. I wonder if this is common problem. Firstly, all of my calls are terminated after 30s. And termination happens in a strange way. My local asterisk server does not see the disconnection, but remote party is disconnected. Basically, I am still on the phone, while remote party was disconnected. When I hang up, I get something like that: Apr 20

Hi guys, i've installed asterisk to handle multiple voip accounts. I've looked at CDR configs, and managed to have cdr-csv files growing after each call. It would be easier to check my locak asterisk cdr's than logging into each account and check them at the provider website. i found that if i ring my sip softphone from my ata, bill seconds are counted correctly. however, if i

I have seen a lot of references to SER. Currently, I have: 1 PRI to Telco 1 PRI to old PBX Several SIP phones with the intention of having approx. 200. I do have people that travel with softphones (currently X-Lite, but will be testing EyeBeam for better codec and echo cancel capabilities) Currently the traveling users have to VPN in first which I am sure is adding extra overhead to the calls. I

Hello everyone! I am using wine 0.9.30 with openSUSE 10.2 I've tried to install and run VoipStunt, and program installs with no error, but fails to start with the following output: dodo@Locutus:~> wine "C:\Program Files\VoipStunt.com\VoipStunt \VoipStunt.exe" preloader: Warning: failed to reserve range 00000000-60000000 err:module:import_dll Library gdiplus.dll (which is needed

Hi, does any know why? i can make call out with my asterisk and voipstunt but i can't get call in on my voip in number i get rejected. if i use Sipura without asterisk i get in calls here is my sip.conf ---------------------------------------------- [general] useragent=nedi port=5060 context=default ;tos=lowdelay disallow=all allow=ulaw allow=alaw allow=gsm allow=g726 language=de

Hi;? When connecting via VoipBuster or VoipStunt, "I can hear them but they can't hear me"?. This happens with VoipBuster or Voipstunt. Registration is done correctly. I thought it could be something related to NAT, but I don't have this problem when using VoipUser or Asterisk2PSTN, for example. ? I tried with different codecs: gsm, alaw and ulaw but no change. ? So, now?I

Hi, guys. May be someone could give me advise? I am trying to find good VOIP provider ONLY for OUTGOING calls with low per channel cost and cheap rates on Eastern Europe, Turky and xUSSR. Should support g729 or g723 codecs, SIP or IAX connectivity. -- ========================================================================= = Best regards, Nikolay Pavlov.

Hi I seem to have installed the G729 codec properly but can't seem to get it to work .. in the Asterisk startup I see .. [codec_g729b.so] => (Annex B (floating point) G.729/PCM16 Codec Translator) == Detected 1 licensed G.729 transcoders WARNING[1074408160]: File translate.c, Line 218 (calc_cost): Translator 'g729tolinb' does not produce sample frames. == Registered translator

Simple as that please email me direct. voipviews@gmail.com Also looking for a U.S. DID provider as well as orig provider.

[Jan 23 19:56:44] -- Got SIP response 300 "Multiple choice" back from 194.120.0.201 [Jan 23 19:56:44] -- Now forwarding SIP/601-fc4d to 'Local/194.120.0.211:5060,sip:194.221.62.211:5060,sip:80.239.235.211:5060@default' (thanks to SIP/voipstunt-5c8c) [Jan 23 19:56:44] NOTICE[3439]: chan_local.c:483 local_alloc: No such extension/context

Hi all..!! I'm developing a system to use festival,asterisk, digium card. This system makes a auto-dialing (using digium TE410P card), when the customer answer the call: asterisk play an audio (pre-built from festival), that is a easy part. our problem is how we can detect a busy line or error in the line but outside of asterisk (by means of a process to detect status of call-line) then we

>I read your postings about faxdetection with CAPI. Interesting feature >but I'm not sure how to implement it on incoming calls (for example fax >dialed wrong number). Can you please show me a dialplan example how you >use this feature (incoming fax calls into Asterisk server). I did'nt test incoming fax, so this is how I would set my conf files : capi.conf : [contr1]

Hi all, Just wondering if someone have already done something like that : SIP Client_A ---> 1)call ---> ASTERISK ---> 2)outgoingcall-PSTN-->Client_B | | 3) Enter conference | MeetMe <----------------------------' with user A Make 2 user in conference (point 1 and 2), it's definitely easy, but call

Hi everybody. We have a samba4 domain deployed across serveral countries. Some of them (overseas) have a poor VPN connection with mainland. Since Samba4 does not support (yet) subtrees, we have deployed a DC in each location for domain validation. However users in mainland logon randomly at overseas location and sometimes this is a problem due to low bandwidth available. Is there any way to

Hi folks, Just wondering if someone have already done something like that : SIP Client_A ---1)call---> ASTERISK ---2)outgoingcall-PSTN-->Client_B | | 3) Enter conference | MeetMe <----------------------------' with user A Make 2 user in conference, it's definitely easy, but call an other user and put the

i would like to carry some oversea pstn-destinations via sip to providers like stanaphone, however, in case of a network-failure or if the provider is not available, i want to fallback to the zap-channels so the call is carried out to the pstn directly. the usual approach would be to check the dialstatus(e.g.NOANSWER). however, asterisk tries >60seconds to reach that peer(even when the ip

I am a newbie to Asterisk , and I am doing research in Asterisk, hope that can get some guidance from the experience users . 1. I wonder Asterisk can do this (refer to the following diagram) or not ? (Can I make a call from the SIP phone to the normal phone ) Asterisk server 1 Asterisk server 2 ======= ======= | |

Hi, what i need is as the example i attached in the box plot. As you can see, the boxplot have margins and it kind of follow the coding I've added, but, it does not happen for the heat map. Could you help me with this.. Thanks in advance. the following is the coding i've used: # Draw the heatmap # heatmap.r # # Purpose: Create a heatmap # # Input: Data matrix as