similar to: # (send immediately) and dialplan broken on PAP2?

Anybody ever run into a case where the Sipura Dial Plan will not work with the S0 option to immediately connect? My Dial plan reads (*xx|[3469]11S0|0|00|[2-9]xxxxxxS0|1xxx[2-9]xxxxxxS0) and I can dial ONLY then numbers in the dial plan so I know that it works. For some reason when I dial 5551212 1212121212 It does not dial for a while and then it dials 555 1212 Anyone have any ideas?

I have a Linksys PAP2. Identical setups for the two channels in both the unit and in Asterisk. In particular, both channels enable g729 and set it as the preferred codec, and have disallow=all and allow=g729 in sip.conf. If we make a call on one channel, it works (and uses g729), but if we make a call on the other channel when the first one is still connected, it fails. We have three g729

Hi, I have "callwaiting=yes" in my zapata.conf, and "Call Waiting Serv: Yes" in the PAP2s. However, there's sitll no callwaiting on the PAP2s. Everything else work fine. Any ideas? Am I missing something somewhere? Thank you. AK -------------- next part -------------- An HTML attachment was scrubbed... URL:

Anyone have an example XML file for the PAP2T? Cheers, j

Hi there, We have plans to install an office (not call center) with the following setup: 200 linksys 942 phones (sip + g711) on a LAN a server with a dual port E1 sangoma and a remora card with 4 fxo modules. So far when we want to setup a linksys phone, we need to use the http interface of each phone, disable/enable a lot of things and plug it into the network. this is not the best scenario for

Is there a way to configure the linksys pap2 to automatically connect to asterisk on lifting the handset (presumably into the 's' state)? Asterisk would then be listening for DTMF tones to figure out what to do rather than having to put a dial plan into each pap2. I think the pap2 is pretty much the same inside as a few of the sipura boxes so the same thing might work if anyone knows...

I receieved my first PAP2-NA yesterday from our distributor(Tech Data). It installed pretty easily and has worked great so I went to order some more of these units today. When I logged into Tech Data this morning, the PAP2-NA was now marked as discontinued and no longer available and only the PAP2 version was available which is the Vonage branded version. :( I saw someone on the list say that

> I have "callwaiting=yes" in my zapata.conf, and "Call Waiting Serv: Yes" > in the PAP2s. > However, there's sitll no callwaiting on the PAP2s. Everything else work > fine. Any ideas? Am I missing something somewhere? Hi Andy, You also need to set "CW Setting: Yes" on the User 1 and User 2 screens. Or, dial from each line, whatever you have set for

Here is some info that may allow some "locked" PAP2 and PAP2-NA units to be used with Asterisk: I have a PAP2-NA (from a provider other than Vonage) for which I did not know the admin password, though the "user" pages were accessible to me. The provider had set it up to fetch at startup, its configuration file by HTTP from a numeric IP. It was running 2.0.10(LSc). A search

I know this is slightly off topic, but I was wondering if anyone can help with a problem getting my PAP2's to connect to Asterisk. I use a provisioning file, and I recently re-wrote the files for each PAP2. I had a small typo and the PAPs logged it as a corrupt file. I corrected the file, however, Line 1 on both of the PAP2's now wont register. Line 2 works fine though. I've done the

Hi all, i need sample xml configuration files for linksys pap2, linksys pap-2t, sipura 2100, sipura 2102, 1001, 3000 and 3102. All of these are linksys/sipura products. So if anyone has these sample files then plz share. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Since the 3000 has been out for a little while, has anyone done a review of the product? (couldn't find anything on google for wiki). Can the fxo and fxs ports be used as two independent channels? Is it really read for prime time? Etc. Rich

Just come across a problem - we have sent out heaps of PAP-2 ATA's and just discovered that when joined in a conference they are choppy on the up leg (so the other users in the conference will hear them with a choppy sound) but the down leg is perfectly fine (so the end user can hear the conference participants perfectly). I have tested the same setup with different brands of ATA's

I just need a "temporary" pppoe server, that only uses PAP ___Can someone post a howto, just in a few lines, what to do?___ If I just do "pppd require-pap" then it gives this error: pppd: The remote system is required to authenticate itself pppd: but I couldn't fin any suitable secret (password) for it to use to do so. pppd is by default installed. and there are just a

I'm trying to set up a Linux-based dialin server on our company network. I'd like to have PPP authenticate using winbindd, if possible. I feel like I've almost gotten it to work, but I can't quite get there. Files: /etc/pam.d/ppp: #%PAM-1.0 auth required pam_nologin.so auth sufficient /lib/security/pam_winbind.so account required

Hi, Did anyone try to set up alarmreceiver application over IP network? Which ATA can I use? I tried to set up it with Linksys PAP-2 but with no luck. Maybe I did something wrong with alarmreceiver.conf (I tried diverse settings, but nothing worked). Sometimes alarmreceiver is able to get some events but sometimes not. I think Linksys PAP-2 has a problem with recognizing digits in appropriate

There is a line in my message that was incorrect. It should state the following: The users with "(primary)" attached to their name are being deleted. As you can see, the Domain Admins group no longer includes joeblow, snoopy or vbe. My Enviornment: NT 4.0 PDC Debian (Sarge) Samba 3.0.7-2 Openldap 2.1.30-3 Windows 98,2000 and XP clients ----- Forwarded by Alex Brown/Compsrv/pap on

Hi all, I'm trying to provision my PAP2T's to use a SVR lookup to find the Asterisk server. I'm using a provisioning file that contains an element like: <Proxy_1_> _sip._udp.example.com </Proxy_1_> However, the PAP doesn't seem to be able to find my server with this hostname. The DNS records are in place because my Polycom and Grandstream servers work just fine.

My goal is to make VPN access to our Samba PDC (FreeBSD 7.0) so that users can access there home shares from Windows clients. I have read the instructions at http://samba.org/ftp/unpacked/lorikeet/pppd/final-report.pdf, but I can't make it work. Don't know if is due to my lack of skills or has something to do with the Popop functionality in FreeBSD. Following the instructions, i made a

(Im' answering cc the list, so the knowledge keeps there, and maybe some more qualified answers become). Am Mittwoch, den 15.10.2008, 18:00 -0700 schrieb Francisco del rosario: > Hey Rodolfo... Need some help from you ... > I need to know what hardware do I need to make SIP calls if I set-up > asterisk > So the situation is that I have a PC and configure the software of my PC to