No Voice When The Calls Come From Internet

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Hi,

I have trouble establishing a call between between two SIP phones. One sip phone is, with asterisk server, at home behind a firewall. The second sip phone is an iPhone with 3G wireless connection.

When I call from the SIP device at home the SIP account on the Internet (iphone + 3G) I can hear voice on both devices. But when I call from the Internet to my home SIP I get the ring but when I answer I don't hear any thing!

I have asterisk v11 installed at home all the incoming TCP/UDP 5060 and a defined UDP…

Asterisk Users 1.4 years ago 1 Answer

CEL For Attented Transfer

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Hi list, I'm trying to use CEL to display channel information in real time. It works fine for simple calls, blind transfers, or SIP attended transfers (initiated from the SIP phone). My problem is for Asterisk attended transfers (atxfer as configured in features.conf). The scenario is: . phone 107 calls phone 100, . 100 dials the atxfer code, . 107 is on hold, and 100 hears the transfer message, . 100 dials phone 103, . 103 answers, . 100 hangups, . 107 and 103 are connected, . 107 hangups. CEL is configured with apps=all and events=ALL, and events are stored…

Asterisk Users 1.8 years ago 7 Answers

Transfer By Transfer Button, Distinguish In Dialplan

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Hi list!

We're using Asterisk in a setup where you can transfer a call via the Asterisk feature, or via the transfer button on a SIP phone. Both work.

However, in my dialplan I cannot distinguish normal calls from calls made by pressing the Transfer button on a phone.

To clarify, I would like to know the difference in the call from A to B, in these two situations:

1 (normal call) A calls B

2 (transferred call) C calls A A presses "Transfer" button and calls B

Does anyone know a way how to accomplish this? Is it at all possible?

Thanks in advance for any…

Asterisk Users 2.6 years ago 0 Answers

Asterisk Realtime Extension... Strange Behaviour

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Hi,

I encountered a strange behaviour using realtime extensions... (on Asterisk 11.2)

when I use the following static dialplan, everything works as expected..:

[from-sip] exten => 110,1,Dial(DAHDI/g0/${EXTEN}) exten => 112,1,Dial(DAHDI/g0/${EXTEN}) exten => _XXX,1,Dial(SIP/${EXTEN}) exten => _X.,1,Dial(DAHDI/g0/${EXTEN})

will say... if a sip phone calls "110" or "112" the call is routed into PSTN (german emergency call) if a sip phone calls any three digit number, the call should be routet to the corresponding SIP user and if a sip phone calls any other number the call should be routed into PSTN... thats ok and works as expected.

when I change to realtime: [from-sip] switch => Realtime

and…

Asterisk Users 2.6 years ago 1 Answer

How Configure Asterisk Server Extension.conf.

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Hi, I have to create scenario like following, I have 2 sip soft phone.I configured Asterisk server on local network, on Linux.With two soft-phone , local asterisk sever, i able to communicate.Now i have communicate with other network SIP client.For that i have opened account at @sip2sip.info, they provided me credentials.Then i registered one SIP phone to local Asterisk sever and another to Sip2sip.info , Can i able to communicate with this scenario ? How i should configure extention.conf in local asterisk sever to communicate with soft-phone which registered at Sip2sip.info ??Or if you have any other idea to crate…

Asterisk Users 2.6 years ago 0 Answers

Odd Cracking With SIP->DAHDI

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We recently set up a SIP trunk between an office in NY running Asterisk and an office in Paris (running Alcatel). All works fine if a SIP phone on the NY system talks to the Paris PBX. But if something on DAHDI (a PRI or MeetMe) talks to the Paris PBX, there's a low-volume crackling. This isn't clipping because it also occurs when there's no legitimate sound. It's sort of a mild version of what you used to get when a POTS pair had a ground short. This occurs no matter what size originates the call.

pings show round trip times…

Asterisk Users 2.9 years ago 5 Answers

Questions About Fax Detection

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Hello,

I want to offer SIP phone user a "custom fax-to-email" feature. Here is how I would describe this feature:

- for every SIP phone,a custom email address is defined - when a SIP phone answers an incoming call (from a trunk or another SIP endpoint), it detects the call is coming from a fax machine and then : + it plays a pre-recorded audio file to the receiving user ("You are now receiving a fax call, please check you email box") + while at the same time, the incoming channel is forwarded to an appropriate statement within Asterisk dialplan. - when…

Asterisk Users 3 years ago 0 Answers