similar to: Asterisk with SER

> Hello fellow asterisk people! > > I have Asterisk listening on port 5061 and SER on port 5060. > > Asterisk is configured as a gateway for ISDN/Analog/H323 and also SIP. > > My problems are with SIP. I can make incoming calls from SIP to asterisk > and to any of the other networks, but when I try to make an outgoing call > from Asterisk to SER I see the following in

Hallo all do any of you know if the toll free access to the Netherlands is still working via FWD or Iaxtel? thanks liaan --------------------------------- Do you Yahoo!? Yahoo! Search presents - Jib Jab's 'Second Term' -------------- next part -------------- An HTML attachment was scrubbed... URL:

I need to setup quota on //server/home for user a to 1GB and on //server/client (for all users on this share) to 10GB May you please let me know the exact command to be used for this? Also, do I need to add any other parameter in smb.conf in order to make quotas work? I have compiled samba with quotas option. Thank you. --------------------------------- Do you Yahoo!? Yahoo! Search presents

Hello, In an attempt to randomise the basic blocks in a function, is it possible that I can randomise the entry block as well? And maybe insert some instructions in the pass to call entry block while running the program ? Is it feasible? What does entry block consist of ? Thanks Tanu --------------------------------- Do you Yahoo!? Yahoo! Search presents - Jib Jab's 'Second

Hello, Is it possible to display the data structure graphs created by the Data Structure Analysis using a visualization tool? Thanks --------------------------------- Do you Yahoo!? Yahoo! Search presents - Jib Jab's 'Second Term' -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have setup a SAMBA + OPENLDAP server following Samba-3 By Example, but I have run into several problems. All the tests described in the Chapter 6 such as pdbedit -Lv, slapcat, and ldapsearch -x -b give the desired results. But, getent can't read passwd or group information from ldap backend. I have done my best to solve the problem, but it just doesn't work. I have compiled the nss_ldap

Using Asterisk 1.0.9 When I try to make an outgoing call with SIP I get the message " 513 Message too big" back from SER. Any ideas what I am doing wrong? Debug below. SER and Asterisk are running on the same Server SER is on port 5060 Asterisk is on port 5061 In my extension.conf I have the line SERADDRESS=192.219.85.57:5060 in Globals and am using exten

Dear All, . After installing, testing and like many other I found that Asterisk is reliable and a great Open source telephony solutions for a professional use. I would like to create a business with *, offering it as a IP PBX solution to customer, as a server, whatever with some Digium HW and SIP phone, .. etc. Asking for something that might have been asked. What are the implications for

Hi all I am in this strange situation: we had ser configured to relay calls to numbers to asterisk extensions and all used to work nicely, with both ser and asterisk running on the same machine with public ip (ser on port 5060 and * on 5061). We had to move temporarily our server to another provider which put our server on a dmz, so that now we have our server with private ip but reachable from

I have been at this problem for almost a week on my own... I am running Samba 3.0.4 on a linux 2.4.26 kernel. I have a share that I want to be read only to the world and writable by members of the write list: [Content] path = /mnt/win/Content read only = yes quest ok = yes write list = sam I set a password for sam using smbpasswd. I believe all of the UNIX permissions are set correctly. I can

hallo all could anyone tell me how to get the * to send keepalive packets over a registration "trunk" or how to increase the amount I'm having natting issues, (the machine is siting behind 2 nat firewalls) thanks liaan --------------------------------- Do you Yahoo!? Yahoo! Search presents - Jib Jab's 'Second Term' -------------- next part -------------- An HTML

Hello fellow asterisk people! I have Asterisk listening on port 5061 and SER on port 5060. Asterisk is configured as a gateway for ISDN/Analog/H323 and also SIP. My problems are with SIP. I can make incoming calls from SIP to asterisk and to any of the other networks, but when I try to make an outgoing call from Asterisk to SER I see the following in Asterisk: -- Executing

Tanu Sharma wrote: > Hello, > > In an attempt to randomise the basic blocks in a function, is it > possible that I can randomise the entry block as well? And maybe insert > some instructions in the pass to call entry block while running the > program ? > > Is it feasible? > > What does entry block consist of ? The entry block, by definition, is the first basic

Hi everybody, I am trying to make up call flow diagrams for for a setup which include ser as a sip proxy/registrar and asteriks as a voicemail server. Is my sequence correct?: UA 1 send an invite to SER. SER forwards this invite to UA2. UA2 sends back a sends back a 100 trying and 180 ringing message. SER forwards these. However UA2 doesnt answer the phone,so what happens then?...is there a

Thanks a lot for replying. I have another query. If branching to the entry block is not legal in llvm how should I be able to create a new entry block for any existing list? I created a new block and inserted it into the present basic block list , but I get this error: opt: BasicBlock.cpp:83: virtual llvm::BasicBlock::~BasicBlock(): Assertion `getParent() == 0 && "BasicBlock

Hello All, I am looking for a stable calling card billing software for Asterisk & find that there are a lot of softwares freely available at http://www.voip-info.org/tiki-index.php?page=Asterisk%20Prepaid%20Applications Does anyone know which is the best of them all or has most features ? Any suggestions ? Suggestions for paid softwares are also welcomed preferably not

Bon soir, * People, You guys are fantastic. This list is hyper-helpful; thanks for that. I have an organization that needs to roll-out * for each of its 30+ locations across the country & we want all features in a DB where they can be easily toggled on & off, as needed. I have done some research in the usual places & got a bit of it, but not all of it, & am hoping someone else

Hi I was trying to use ChanSpy command, but it seems like it is not implemented, or is not included in the standard asterisk distribution. Can someone tell me how to obtain this. I am trying to monitor IAX channels, and I know now that ZapMonitor cmd doesnt work on this,so can anyone tell me how to monitor IAX channels. Thanks in advance --------------------------------- Do you

I have a question regarding to OS platform. As I see on Wiki -s homepage there are many type of linux version. And in some of them there are reported errors (regarding to asterisk ) for exemole in rad hat . Can you tell me what is the best linux paltform ,( version ), which is supported by digiroom card (T1 and TMD )and asterisk run on it stable ?. Which linux is prefereable ? for asterisk ?

Hi all I installed a tdm400p into a old p2 machine. I'm not able to see it under /proc/interupts or using lspci.. we removed all other cards. changed slots, forced irq to that slot.. etc etc. what is the min specs needed to get one of these cards running? thanks L --------------------------------- Do you Yahoo!? Yahoo! Search presents - Jib Jab's 'Second Term'