Hi, i have a sip trunk between Asterisk and SBC Huawey 2900 , i try to send with user=phone in From, and try to use Sip headers like PAI and PID and nohing happended, the Caller ID its always “unknow”, how can set and pass our caller id, thanks
Please provide the relevant parts of pjsip.conf, and the “pjsip set logger on” output.
If you are using chan_sip, please change to chan_pjsip and retry.
“unknow” doesn’t sound right. I’d expect a full English word.
The word “trunk” doesn’t appear in the SIP RFC.
thanks david551
witch version of asterisk are you using
hi Mark im using Asterisk 11 (Issabel distro), i try everithing, but the SBC Huawei don´t recognize the caller id, the log of the Huawei says “error 400” CAll-ID : anonymous@anonymous Callid header absent or undecipherable"
Asterisk 11 went EOL back in 2017
please upgrade to Asterisk 20 or atleast 18
also have you checke the Issabel forum if they can help with configureing
as we have no idear how to configure the web gui for Issabel
Call-ID is not Caller-ID.
The Call-ID header shown is syntactically valid, although semantically invalid in a way that Asterisk would never do (it must be different on every call). The capitalisation is strange, but appears to be acceptable, and the space before the colon is also acceptable.
Personally I don’t know anyone using Asterisk 11 anymore. Some still use 13, but you should consider an upgrade for sure!
yep that is fucked up
can you dig up the INVITE send to Huawei from Asterisk (on asterisk)
also pleas do not post pictures, post it as preformatted text
Sorry i have only this pictures because my customer have the acces to the issabel machine
hmm that one look normal
is there anything between Asterisk and Huawei that could interfeer like a router/firewall (broken sip alg)
before asterisk there was a kamailio and with the same ip, but with it there was no call id problem but there was an audio problem, now we have issabel without audio problems but with callerid problems
That’s the response. The Call-ID it is objecting to is in the request. I suspect it has put anonymous@anonymous just to make something that is syntactically valid
If this were chan_pjsip, I could speculate that the HuaWei doesn’t like call-IDs without an @ which are perfectly valid. However, although chan_pjsip would be possible, if you are locked into Asterisk 11, there is a good chance you are locked into chan_sip. which does use @.
Note that I don’t know what anything but the plain text in the Warning means. That is determined by the HuaWei.
It’s not a caller ID problem; it is Call-ID problem. The former is the identity of the caller and the latter of the call. Think of a formal business order. The caller ID is the customer account name, but the call ID is the order number. Although invoices are not quite as good a fit, call ID is like the invoice number.
can you check tht SIP ALG is disabled
Breaking SIP signalling: Many of the actual common routers with inbuilt SIP ALG modify SIP headers and the SDP body incorrectly, breaking SIP and making communication just impossible. Some of them do a whole replacing by searching a private address in all SIP headers and body and replacing them with the router public mapped address (for example, replacing the private address if it appears in “Call-ID” header, which makes no sense at all). Many SIP ALG routers corrupt the SIP message when writing into it (i.e. missed semi-colon “;” in header parameters). Writing incorrect port values greater than 65536 is also common in many of these routers.
also if posible get the INVITE that the Huawie is reciving just to check that nothing has altered the INVITE that Asterisk send
hello everyone, last night we tested going back with the server that was before asterisk, which was a kamailio, and we tried altering the from field but the caller id passes without problems, so the only difference we see between what kamailio sends and what asterisk sends is the length of the “Call-ID” field since the length in Kamailio is shorter than in asterisk which exceeds 15 digits, there is some way to modify this in Asterisk?
hmm I belive the max length is 256 characters
the only way to modify it would be changing the code
also try switch to PJSIP as its format for CALL-ID is diffrent and may be motr to Huawie liking
can you check if there exist a FW update for Huawie