similar to: configuring ser for *

Any more ideas on my below mail? If a user is registered with SER and leaves a voicemail message with asterisk (by using rewritehostport etc in ser.cfg), then how is the user supposed to listen to the message afterwards? Is there any other way other than the MWI method?? Thnaksm Aisling. ---- Original Message ---- From: ashling.odriscoll@cit.ie To: asterisk-users@lists.digium.com Subject: FW:

Just another bit of info which might help solve this: Looking at the Asterisk log messages - I notice when I start up Asterisk, I see the error: pbx_config.c: Can't use 'next' priority on the first entry! Could I be right that its something got to do with priorities? I changed the incomingpstn context to the following eliminating the 'n' field and still the same errors were

Hi, Yes InternalExtension is the context and 2093 the extension. Just to explain something odd that?s happening (and I?m very stumped with this) .I think my contexts are definately the reason that I can?t interrupt the menu for incoming pstn calls to choose a submenu: My users register with my sip proxy (SER). Therefore when I create an entry for them in sip.conf I set only one context. Also to

Hello, I think I might have an inkling as to where the issue may be at. For some reason when I create a new context, a directory is not created in /var/spool/asterisk/voicemail. The default context resides there but new ones are not created. Has anyone ever experienced this or does anyone have any idea as to how I would solve this? Hope someone can shed light on this, Many thanks, Aisling.

Following on from my below email, things have taken another bizarre twist.. I have continued getting the error when 2092 tries to listen to messages by dialing 9999. --Executing VoiceMailMain ("SIP/2092-6918", "2092") in new stack --Playing 'vm-password' (language 'en') WARNING: app_voicemail.c:4922 vm_authentication: Unable to read password. Then I

Hi all, Hope somebody can help-I really am stumped as to why this won't work. I realise that this isnt an Asterisk problem (Please dont bash me on the list) and I have emailed the SER list but I havent received a reply and maybe someone on this list can help...Once this problem is solved I am going to use Asterisk for voicemail etc with SER (I have it set up) I currently have SER set up and

Hi all, I have SER and Asterisk set up together with ser handling user registrations and asterisk providing voicemail services. When I ring a phone and it doesnt answer after a designated amount of time, the request is forwarded to asterisk, and I can leave a message. Now, this may seem a ridiculous question but how can I listen to my message afterwards? I have read about a solution by Java

Hello, Hope somebody can help me - Asterisk is behaving very oddly and I'm totally stumped! I have SER and Asterisk running on the same box. I want SER to listen on port 5060 (it is) and Asterisk to listen on port 5062. I have configured my phones to register with x.x.x.x:5060 (SER) and Asterisk will purely act as a voicemail server at the moment. However I cannot get Asterisk to listen on a

Hi, I am getting the following error when I attempt to listen to voice messages by dialing 9999 (I can hear nothing): --Executing VoiceMailMain ("SIP/2092-6918", "2092") in new stack --Playing 'vm-password' (language 'en') WARNING: app_voicemail.c:4922 vm_authentication: Unable to read password. I read in previous posts that this may be to do with the dtmf

Hi, First of all apologies because this isn't strictly a purely asterisk question. I am quite new to asterisk and actually to voip/telephony as a whole. I currently have sip calls working through asterisk. The asterisk server is behind a linksys router. I would now like to connect calls to the pstn. I have researched into several ways to do this but because I am not very knowledgeable about

Hi all, I am having difficulty getting incoming PSTN calls working. I have set up an account with a third party provider. In my system, the user register with SER and use Asterisk for PSTN access, voicemail etc My provider told me to change my sip.conf as follows register => username:password@sip.blueface.ie/2093 ; To receive incoming calls specify this block and

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

Guess I got where your confusiion lies.... you DON'T set up the ALL contexts the users will have acces to in sip.conf, in this file you will only set 1 single context. All other contexts you want the users to have acces to, you will have to add them in the extensions.conf So what you have to do is the following: -user 2092, set it the createmenu context in sip .conf - in extensions.conf

Hi, I am trying to update Asterisk from cvs as I think it might solve a secondary problem that I am experiencing (see below). In the /usr/src/asterisk directory I typed "make upgrade". However I get an error: Makefile:16: *** missing separator. Stop. Make[2]L Leaving directory '/usr/src/asterisk' Make: *** [depend] Error 1 Has anyone come across this or does anyone know a

Hi all, Could someone let me know the most common way that an Internet ISP would allow customers access to the PSTN?? Do they buy multiple fxo cards such as the TDM400P and rent multiple lines from a larger provider?? Would the best way be to connect to a third party voice/pstn gateway?? Is that simply a matter of forwarding all sip traffic destined for the pstn to another provider with a

Hi, I'm wondering if anyone has experienced an issue with the XLite softphone and asterisk accepting dtmf? I can listen to my voicemail perfectly from my hardphone. However when I dial the voicemail number from my XLite softphone and enter the password at the voicemail prompt, an error appears vm-incorrect and I get an "Unable to read password" message on the asterisk console. Has

Hello: I have successfully install spandsp and patch asterisk with it. But when I received a Fax is garble or shrink. Does any one know why???... Am using a PRI T100P card to receive the fax and save it to a tiff file... Any help will be greatly appreciated. Here are the versions. Latest csv from asterisk, spandsp-0.0.1k.tar.gz redhat 7.3 T100P has its own IRQ. Any help will be greatly

Hi guys im trying to get NAT working on my system. im using 3 phones, 2000 = xlite, 2001 = xlite, and 2010 = some piece of crap voip phone. when i ring from anywhere to anywhere u can either never hear anything on both ends, or just 1 end can hear stuff. below is the output of sip show peers 2010/2010 203.1.68.90 (D) 255.255.255.255 49534 UNREACHABLE 2001/2001 203.1.68.90

I wonder does anyone have any thoughts or can give me some direction on the following: I have an asterisk testbed environment set up. My task is to make a personal number service available whereby users would be given one number (perhaps a voip number) and this number would enable them to be reached via the pstn, pots, gsm etc.... Does anyone have ideas where I could start looking at sites to

Why is SER considered a better SIPserver than asterisk , why is it that SER can handle more clients than asterisk can. And if this is just cause of say poor SIP handling code in asterisk then is there anything being done to fix it. Just wanted to know why SER claims to be better than asterisk as a SIP server. ? -- regards Vikram (http://www.vicramresearch.com)