* You are viewing Posts Tagged ‘sip phone’

No Voice When The Calls Come From Internet


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 range in rtp.conf forwarded to my Asterisk server.

Do you have any idea when the voice is heard only when the call is from my local network to the Internet and not in the other direction ?

Nevertheless, when both SIP devices are in the same home IP network the call is made without any problem whatever who starts the call.


Dialplan To Reach External SIP Phone

If I have Asterisk setup with local SIP phones configured but need to call a SIP phone which is not local but actually on another VLAN, what would the dialplan need to look like?

CEL For Attented Transfer

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 in a database via cel_pgsql.

This is the list of events in the database for this call:

eventtype | channame | peer

CHAN_START | SIP/107-0274 |
CHAN_START | SIP/100-0275 |
ANSWER | SIP/100-0275 |
ANSWER | SIP/107-0274 |
BRIDGE_START | SIP/107-0274 | SIP/100-0275
CHAN_START | Local/103@100-0042;1 |
CHAN_START | Local/103@100-0042;2 |
CHAN_START | SIP/103-0276 |
ANSWER | SIP/103-0276 |
ANSWER | Local/103@100-0042;2 |
BRIDGE_START | Local/103@100-0042;2 | SIP/103-0276
ANSWER | Local/103@100-0042;1 |
BRIDGE_START | SIP/100-0275 | Local/103@100-0042;1
BRIDGE_END | SIP/100-0275 | Local/103@100-0042;1
ATTENDEDTRANSFER | SIP/107-0274 | Local/103@100-0042;1
CHAN_START | Transfered/SIP/107-0274 |
BRIDGE_END | Transfered/SIP/107-0274| SIP/100-0275
BRIDGE_START | SIP/107-0274 | Local/103@100-0042;1
HANGUP | SIP/100-0275 |
CHAN_END | SIP/100-0275 |
HANGUP | Transfered/SIP/107-0274|
CHAN_END | Transfered/SIP/107-0274|
BRIDGE_END | SIP/107-0274 | Local/103@100-0042;1
HANGUP | Local/103@100-0042;1 |
CHAN_END | Local/103@100-0042;1 |
HANGUP | SIP/107-0274 |
CHAN_END | SIP/107-0274 |
BRIDGE_END | Local/103@100-0042;2 | SIP/103-0276
HANGUP | SIP/103-0276 |
CHAN_END | SIP/103-0276 |
HANGUP | Local/103@100-0042;2 |
CHAN_END | Local/103@100-0042;2 |
LINKEDID_END | Local/103@100-0042;2 |
(33 lignes)

How should these events be interpreted?

Asterisk version is 11.6.0.

– —
Jean-Denis Girard

SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27

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OT – Question On Aastra 6735i – Was: Sip Phone Displaying Caller Name While On Call


2013/4/19 Olivier

I’ve been testing Aastra 6757i SIP phone and it appears that this phone has the specific feature I was looking for : displaying both remote party name and number while on call.

Is this “feature” also available in Aastra 6735i ?


Transfer By Transfer Button, Distinguish In Dialplan

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 hints,


Asterisk Realtime Extension… Strange Behaviour


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

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

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:
switch => Realtime

and put the diaplan into the database id context exten priority app appdata
“1” “from-sip” “110” “1” “Dial” “DAHDI/g0/${EXTEN}”
“2” “from-sip” “112” “1” “Dial” “DAHDI/g0/${EXTEN}”
“3” “from-sip” “_XXX” “1” “Dial” “SIP/${EXTEN}”
“4” “from-sip” “_X.” “1” “Dial” “DAHDI/g0/${EXTEN}”

only the emergency calls work and any other call goes to DAHDI… I cant reach any other SIP phone. Even when swapping the content of the rows 3 and 4 in the database to id context exten priority app appdata
“1” “from-sip” “110” “1” “Dial” “DAHDI/g0/${EXTEN}”
“2” “from-sip” “112” “1” “Dial” “DAHDI/g0/${EXTEN}”
“3” “from-sip” “_X.” “1” “Dial” “DAHDI/g0/${EXTEN}”
“4” “from-sip” “_XXX” “1” “Dial” “SIP/${EXTEN}”

makes no difference… I thought, using realtime extensions would read the dialplan from top to bottom, ordered by “id”… but it seems to be ignored somehow and the extension “_X.” catches the calls before the extensionpattern “_XXX” is reached.

I _could_ avoid this be prefixing “external” numbers with a leading 0
for example… but I dont want to… as I said.. using static extension via extensions.conf the dialplan works as expected…

Am I missing something?

regards, yves