similar to: early media

Hi, I have asterisk 1.4.16 behind a NAT-FW which is using a hosted SIP trunk for PSTN calling. Asterisk is configured to support nat with nat=yes in sip.conf. Now the hosted PSTN Gateway supports symmetric RTP and early media using 183 Session Progress. So If I call a PSTN number which has IVR message played before the call is connected (via 183), those media RTP packets do not reach the

Hello, I have a problem with the following: When I dial a PSTN number from a UAC, the call is made through a SIP Trunk (which has a connection to the PSTN) in Asterisk. The PSTN Gateway returns a 180 Ringing WITH SDP, but Asterisk forwards the 100 Ringing WITHOUT SDP: As you can see below, the SIP message from 10.254.254.1 (the PSTN Gateway) has SDP, while * (with 192.168.0.173) removes the SDP

Hello, I have a problem with the following: When I dial a PSTN number from a UAC, the call is made through a SIP Trunk (which has a connection to the PSTN) in Asterisk. The PSTN Gateway returns a 100 Ringing WITH SDP, but Asterisk forwards the 100 Ringing WITHOUT SDP: As you can see below, the SIP message from 10.254.254.1 (the PSTN Gateway) has SDP, while * (with 192.168.0.173) removes the SDP

Greetings, upgrading from 1.4.17 to 1.4.19 some asterisk gateway of ours (used for gatewaying ISDN-PRI and SIP), I noticed an annoying thing: when the PSTN party answers, for a few seconds (4/5 sec typical) some SIP client could not hear anything (the ringing was heard well!), then the audio comes back again with no problem. Looking for any differences between the behaviour of version 1.4.17 and

Hello all, I've been playing around with early audio, and I'm able to get some things working We have PSTN calls coming in to asterisk in SIP from a Cisco AS5300. If I do the following: Exten => i,1,Playback(ss-noservice,noanswer) Exten => i,2,Congestion(15) Exten => i,3,Hangup() The PSTN caller does not get an answered call (doesn't get billed) but hears the ss-noservice

Disclaimer: Yes, I know faxing over G711 is unreliable. :-) We're running Asterisk 1.0.9 which talks to a Audiocodes SIP Gateway. We're running Sipura SPA-2002's as ATA's and faxing within our own voice network is working. If we try and fax out to the world however, we're running into a problem. When the call connects and the modem tones begin to negotiate, our SIP/PSTN

Hello, Throwing calls from Asterisk to PSTN (via a VoIP gateway) some operators sends an explaining audio, in situations as: The phone number does is not assigned The phone is powered off etc. The audio is sent before the call to be answered. So, in an automatic dialling application I'd like to recognize that audio to know what to do with those calls (queue them to a service, mark as wrong

I am setting up Asterisk for use in a low bandwidth environment. As bandwidth is precious and our ATA's support it, the decision was made to use the g723 codec. I have been working on this for a few days and have not been successful. The issue that I am having is garbled noise at the client on calls whose RTP streams are terminated by Asterisk system. This is the case for all the dialplan

Hi list, I'm evaluating a private telephony scenario of about 20 locations - 300 phones, 50 FAX machines. Initial overview points to the installation of asterisk at three locations connected to the PSTN via ISDN PRI. All other locations, small by themselves, would get SIP phones managed by asterisk, since there is good IP connectivity between all sites. Now on to the

I wanted to get everyone's opinion on an issue I am having. I am currently using linksys PAP2NA ATA adapters to terminate analog calls from my auto dialer to the voip termination co. The problem I have is when I call the PSTN everything goes fine until the person being called hangs up the phone. Once they hang up on the PSTN side it takes almost 15-20 Seconds for the ata to see the

When we make calls out of asterisk to the PSTN via a SIP termination service provider the called party gets a slight echo of their voice. Here is the setup; analog phone <> Linksys ata <> asterisk <> sip provider sonus GSX 9000 <> PSTN <> called party. The caller on the analog phone connected to the ATA hears no echo at all. The called party has a slight

Hi I know that this topic was on the list maybe dozen of times. But I have a question regarding the fax support in asterisk, because all the information I could get does not give me the clear view of if. I read that Asterisk 1.8 will have strong fax (t.38) support, but I want to know if these four scenarios will be possible to achieve: fax machine (phone+fax) connected to ATA --- SPA2102 ATA ---

Apologies if this is an FAQ, I wasn't able to find an answer googling: Will any of the public PSTN/VoIP gateway services (Vonage, Packet8 etc) interoperate with * ? I'd like to deploy a box which provides PBX service for analog handsets, and handles inbound/outbound calls via both analog PSTN lines, and, say Packet8 VoIP service. I understand that I can do this by connecting the analog

Hi there! We have an old legacy norstar phone system m8x24-ds ( dr5 ) and a couple of m0x16. It has 5 external analog lines. It has no auto attendant, and no voicemail. So every incoming call is forwarded to a operator, she pick up the phone, talks to the caller and transfer the call to the right extension. We are in Argentina, so buying a star talk is out of the question, there is no selling of

Here is my Configuration PSTN -> Cisco AS5350 -> SIP -> ASTERISK -> SIP -> ATA186 When I call from the pstn to the ATA, the ATA rings but I don't hear anything on the calling side until the call is picked up. When I call from the ATA, everything seems to work fine. When I bypassed ASTERISK, everything seems to work fine. Anyone know what I might have configured wrong?

Hi, I'm looking for a few pointers on using ATA to connect Asterisk to the PSTN. Basically, I'm running a Hosted PBX service, and in urban centers I can usually get SIP or PRIs. Since I sell my customers SIP hardphones, the data flow is like this: Customer's SIP Hardphone ---- My own Asterisk ----- Outside lines But when it comes to smaller villages (I deal with people in tiny

(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

Hi I've spent two days trying to solve this issue but to no prevail and I'm hoping to get some help. I've configured Asterisk as a SIP client, running on OpenWRT on an embedded device with onboard FXS and ATA. Asterisk is connecting to an external SIP provider on the Internet who in turn provides a PSTN gateway. I'm able to make calls to other SIP accounts registered on the same

Hi, I have two DECT cordless phones (one Philips Onis 6311 and one Philips Onis2 Memo-6511). The first one is for the french standard and the second one for the british standard. Both of them have callerid functionality. The british one does not show anything when connected to my PSTN line. I have not yet tested the french one. What I want to do is to connect both of them to the same ATA box.

To provide a reliable fax solution for users connected to a Asterisk 1.6.2.6 server i have tested a few T.38 capable ATA's: - Patton M-ATA - Grandstream HandyTone 486 - Fritz!Box 7170 I have tried Asterisk 1.6.2.6 compiled with SpanDSP-0.0.6pre17 and also Asterisk 1.6.2.6 with Fax for Asterisk installed. These Asterisk servers are connected to a Cisco PGW 2200 + AS5400XM. Sending fax